Special Publication 

800-58 

Security Considerations for 

Voice Over IP Systems 

Recommendations of the National Institute 

of Standards and Technology 

D. Richard Kuhn, Thomas J. Walsh, Steffen Fries
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

II 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

NIST Special Publication 800-58

Security Considerations 

for 

Voice Over IP Systems 

Recommendations of the National 

Institute of Standards and Technology

III 

C O M P U T E R S E C U R I T Y

Computer Security Division 

Information Technology Laboratory 

National Institute of Standards and Technology

Gaithersburg, MD 20899-8930 

January 2005 

U.S. Department of Commerce

Donald L. Evans, Secretary 

Technology Administration

Phillip J. Bond, Under Secretary for Technology

National Institute of Standards and Technology

Shashi Phoha, Director
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

Note to Readers 

The Information Technology Laboratory (ITL) at the National Institute of Standards and Technology

(NIST) promotes the U.S. economy and public welfare by providing technical leadership for the

Nation’s measurement and standards infrastructure. ITL develops tests, test methods, reference data, 

proof of concept implementations, and technical analysis to advance the development and productive

use of information technology. ITL’s responsibilities include the development of technical, physical, 

administrative, and management standards and guidelines for the cost-effective security and privacy of

sensitive unclassified information in Federal computer systems. 

This document is a publication of the National Institute of Standards and Technology (NIST) and 

is not subject to U.S. copyright. Certain commercial entities, equipment, or materials may be 

identified in this document in order to describe an experimental procedure or concept adequately. 

Such identification is not intended to imply recommendation or endorsement by the National 

Institute of Standards and Technology, nor is it intended to imply that the entities, materials, or 

equipment are necessarily the best available for the purpose. 

T.J. Walsh and D.R. Kuhn are employees of NIST; S. Fries is an employee of Siemens AG. 

For questions or comments on this document, contact sp800-58@nist.gov.

Acknowledgments

This document has benefited from review and comment by many experts. We particularly

want to thank Mike Stauffer, of Booz Allen Hamilton, for his careful review and many 

contributions to improving the quality of this publication. Appendix A is derived from an 

internal NIST report by Tony Meehan and Tyler Moore of the University of Tulsa. Many

people provided us with helpul comments and suggestions for improvements to the first draft

of this document. We are grateful to Tim Grance, John Larson and colleagues at Sprint,

Stephen Whitlock, David Waring, Steven Ungar, Larry L. Brock, Ron Rice, John Dabnor, 

Susan Landau, Cynthia Des Lauriers, Victor Marshall, Nora Wheeler, Anthony Smith, Curt 

Barker, Gerald Maguire, Frank Derks, Ben Halpert, Elaine Starkey, William Ryberg, Loraine 

Beyer, Terry Sherald, Gill Williams, Roberta Durant, Adrian Gardner, Rich Graveman, David 

Harrity, Lakshminath Dondeti, Mary Barnes, Cedric Aoun, Mike Lee, Paul Simmons, Marcus 

Leech, Paul Knight, Ken van Wyk, Manuel Vexler, and John Kelsey.

IV 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

Table of Contents

Executive Summary and Recommendations.......................................................................................... 3

1 Introduction ...................................................................................................................................... 10

1.1 Authority .................................................................................................................................... 10

1.2 Document Scope and Purpose .................................................................................................. 10

1.3 Audience and Assumptions....................................................................................................... 11

1.4 Document Organization ............................................................................................................ 11

2 Overview of VOIP............................................................................................................................ 13

2.1 VOIP Equipment ....................................................................................................................... 13

2.2 Overview of VOIP Data Handling ........................................................................................... 14

2.3 Cost............................................................................................................................................. 16

2.4 Speed and Quality...................................................................................................................... 16

2.5 Privacy and Legal Issues with VOIP........................................................................................ 17

2.6 VOIP Security Issues................................................................................................................. 17

3 Quality of Service Issues.................................................................................................................. 19

3.1 Latency....................................................................................................................................... 19

3.2 Jitter............................................................................................................................................ 20

3.3 Packet Loss................................................................................................................................ 21

3.4 Bandwidth & Effective Bandwidth .......................................................................................... 22

3.5 The Need for Speed................................................................................................................... 24

3.6 Power Failure and Backup Systems ......................................................................................... 24

3.7 Quality of Service Implications for Security............................................................................ 25

4 H.323................................................................................................................................................... 26

4.1 H.323 Architecture .................................................................................................................... 26

4.2 H.235 Security Profiles.............................................................................................................. 28

4.2.1 H.235v2.................................................................................................................................. 29

4.2.2 H.235v3.................................................................................................................................. 32

4.2.3 H.323 Annex J ....................................................................................................................... 36

4.2.4 H.323 Security Issues............................................................................................................ 36

4.3 Encryption Issues and Performance.......................................................................................... 37

5 SIP....................................................................................................................................................... 39

5.1 SIP Architecture......................................................................................................................... 39

5.2 Existing Security Features within the SIP Protocol ................................................................. 40

5.2.1 Authentication of Signaling Data using HTTP Digest Authentication............................... 41

5.2.2 S/MIME Usage within SIP.................................................................................................... 41

5.2.3 Confidentiality of Media Data.............................................................................................. 41

5.2.4 TLS usage within SIP............................................................................................................ 42

5.2.5 IPsec usage within SIP.......................................................................................................... 42

5.2.6 Security Enhancements for SIP ............................................................................................ 42

5.2.7 SIP Security Issues................................................................................................................ 44

6 Gateway Decomposition.................................................................................................................. 47

6.1 MGCP ........................................................................................................................................ 47

6.1.1 Overview................................................................................................................................ 47

V 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

6.1.2 System Architecture............................................................................................................... 48

6.1.3 Security Considerations........................................................................................................ 48

6.2 Megaco/H.248 ........................................................................................................................... 49

6.2.1 Overview................................................................................................................................ 49

6.2.2 System Architecture............................................................................................................... 49

6.2.3 Security Considerations........................................................................................................ 50

7 Firewalls, Address Translation, and Call Establishment........................................................... 52

7.1 Firewalls..................................................................................................................................... 52

7.1.1 Stateful Firewalls................................................................................................................... 53

7.1.2 VOIP specific Firewall Needs .............................................................................................. 53

7.2 Network Address Translation ................................................................................................... 54

7.3 Firewalls, NATs, and VOIP Issues........................................................................................... 56

7.3.1 Incoming Calls....................................................................................................................... 56

7.3.2 Effects on QoS ....................................................................................................................... 57

7.3.3 Firewalls and NATs............................................................................................................... 57

7.4 Call Setup Considerations with NATs and Firewalls.............................................................. 58

7.4.1 Application Level Gateways................................................................................................. 59

7.4.2 Middlebox Solutions.............................................................................................................. 59

7.4.3 Session Border Controllers................................................................................................... 60

7.5 Mechanisms to solve the NAT problem................................................................................... 61

7.6 Virtual Private Networks and Firewalls.................................................................................... 62

8 Encryption & IPsec.......................................................................................................................... 63

8.1 IPsec ........................................................................................................................................... 63

8.2 The Role of IPsec in VOIP........................................................................................................ 65

8.3 Local VPN Tunnels................................................................................................................... 65

8.4 Difficulties Arising from VOIPsec ........................................................................................... 65

8.5 Encryption / Decryption Latency.............................................................................................. 66

8.6 Scheduling and the Lack of QoS in the Crypto-Engine........................................................... 67

8.7 Expanded Packet Size................................................................................................................ 68

8.8 IPsec and NAT Incompatibility ................................................................................................ 68

9 Solutions to the VOIPsec Issues..................................................................................................... 69

9.1 Encryption at the End Points..................................................................................................... 69

9.2 Secure Real Time Protocol (SRTP).......................................................................................... 69

9.3 Key Management for SRTP – MIKEY.................................................................................... 71

9.4 Better Scheduling Schemes....................................................................................................... 72

9.5 Compression of Packet Size...................................................................................................... 72

9.6 Resolving NAT/IPsec Incompatibilities................................................................................... 73

10 Planning for VOIP Deployment..................................................................................................... 75

References .................................................................................................................................................. 78

A Appendix: VOIP Risks, Threats, and Vulnerabilities.............................................................. 81

A.1 Confidentiality and Privacy....................................................................................................... 81

A.2 Integrity Issues........................................................................................................................... 83

A.3 Availability and Denial of Service............................................................................................ 85

B Appendix: VOIP Frequently Asked Questions........................................................................... 88

C Appendix: VOIP Terms................................................................................................................. 91

1 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

Index............................................................................................................................................................ 93

List of Figures and Tables

Figure 1. Voice Data Processing in a VOIP System. ................................................................... 15 

Figure 2. Sample Latency Budget ................................................................................................ 20 

Figure 3. H.323 Architecture........................................................................................................ 26 

Figure 4. H.323 Call Setup Process.............................................................................................. 27 

Table 1: H235v2 Annex D - Baseline Security Profile ................................................................. 29 

Table 2: H235v2 Annex E – Signature Security Profile ............................................................... 30 

Table 3: H235v2 - Voice Encryption Option ................................................................................ 31 

Table 4: H235v2 Annex F – Hybrid Security Profile.................................................................... 32 

Table 5: H235v3 Annex D - Baseline Security Profile ................................................................. 33 

Figure 5. SIP Network Architecture............................................................................................. 40 

Figure 6. SIP Protocol .................................................................................................................. 46 

Figure 7: General Scenario for MGCP Usage ............................................................................... 48 

Figure 8: General Scenario for MEGACO/H.248 Usage .............................................................. 50 

Figure 9. IP Telephones Behind NAT and Firewall..................................................................... 55 

Figure 10. Middlebox Communications Scenario........................................................................ 60 

Figure 11. IPsec Tunnel and Transport Modes............................................................................. 64 

2 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

EXECUTIVE SUMMARY AND RECOMMENDATIONS

Voice over IP – the transmission of voice over packet-switched IP networks – is 

one of the most important emerging trends in telecommunications. As with many 

new technologies, VOIP introduces both security risks and opportunities. VOIP 

has a very different architecture than traditional circuit-based telephony, and these 

differences result in significant security issues. Lower cost and greater flexibility 

are among the promises of VOIP for the enterprise, but VOIP should not be 

installed without careful consideration of the security problems introduced. 

Administrators may mistakenly assume that since digitized voice travels in 

packets, they can simply plug VOIP components into their already-secured 

networks and remain secure. However, the process is not that simple. This 

publication explains the challenges of VOIP security for agency and commercial 

users of VOIP, and outlines steps needed to help secure an organization’s VOIP 

network. VOIP security considerations for the public switched telephone network 

(PSTN) are largely outside the scope of this document. 

VOIP systems take a wide variety of forms, including traditional telephone 

handsets, conferencing units, and mobile units. In addition to end-user 

equipment, VOIP systems include a variety of other components, including call 

processors/call managers, gateways, routers, firewalls, and protocols. Most of 

these components have counterparts used in data networks, but the performance 

demands of VOIP mean that ordinary network software and hardware must be 

supplemented with special VOIP components. Not only does VOIP require 

higher performance than most data systems, critical services, such as Emergency 

911 must be accommodated. One of the main sources of confusion for those new 

to VOIP is the (natural) assumption that because digitized voice travels in packets 

just like other data, existing network architectures and tools can be used without 

change. However, VOIP adds a number of complications to existing network 

technology, and these problems are magnified by security considerations. 

Quality of Service (QoS) is fundamental to the operation of a VOIP network that 

meets users’ quality expectations. However, the implementation of various 

security measures can cause a marked deterioration in QoS. These complications 

range from firewalls delaying or blocking call setups to encryption-produced 

latency and delay variation (jitter). Because of the time-critical nature of VOIP, 

and its low tolerance for disruption and packet loss, many security measures 

implemented in traditional data networks are simply not applicable to VOIP in 

their current form; firewalls, intrusion detection systems, and other components 

must be specialized for VOIP. Current VOIP systems use either a proprietary 

protocol, or one of two standards, H.323 and the Session Initiation Protocol (SIP). 

Although SIP seems to be gaining in popularity, neither of these protocols has 

become dominant in the market yet, so it often makes sense to incorporate 

components that can support both. In addition to SIP and H.323 there are also two 

further standards, media gateway control protocol (MGCP) and Megaco/H.248, 

3 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

which may be used in large deployments for gateway decomposition. These 

standards may be used to ease message handling with media gateways, or on the 

other hand they can easily be used to implement terminals without any 

intelligence, similar to today’s phones connected to a PBX using a stimulus 

protocol. 

Packet networks depend for their successful operation on a large number of 

configurable parameters: IP and MAC (physical) addresses of voice terminals, 

addresses of routers and firewalls, and VOIP specific software such as call 

processing components (call managers) and other programs used to place and 

route calls. Many of these network parameters are established dynamically every 

time network components are restarted, or when a VOIP telephone is restarted or 

added to the network. Because there are so many places in a network with 

dynamically configurable parameters, intruders have a wide array of potentially 

vulnerable points to attack [1]. 

Firewalls are a staple of security in today’s IP networks. Whether protecting a 

LAN or WAN, encapsulating a DMZ, or just protecting a single computer, a 

firewall is usually the first line of defense against would be attackers. Firewalls 

work by blocking traffic deemed to be invasive, intrusive, or just plain malicious 

from flowing through them. Acceptable traffic is determined by a set of rules 

programmed into the firewall by the network administrator. The introduction of 

firewalls to the VOIP network complicates several aspects of VOIP, most notably 

dynamic port trafficking and call setup procedures. 

Network Address Translation (NAT) is a powerful tool that can be used to hide 

internal network addresses and enable several endpoints within a LAN to use the 

same (external) IP address. The benefits of NATs come at a price. For one 

thing, an attempt to make a call into the network becomes very complex when a 

NAT is introduced. The situation is somewhat similar to an office building where 

mail is addressed with employees’ names and the building address, but internal 

addressing is handled by the company mailroom. There are also several issues 

associated with the transmission of voice data across the NAT, including an 

incompatibility with IPsec. Although the use of NATs may be reduced as IPv6 is 

adopted, they will remain a common component in networks for years to come, so 

VOIP systems must deal with the complexities of NATs. 

 

Firewalls, gateways, and other such devices can also help keep intruders from

compromising a network. However, firewalls are no defense against an internal 

hacker. Another layer of defense is necessary at the protocol level to protect the 

voice traffic. In VOIP, as in data networks, this can be accomplished by 

encrypting the packets at the IP level using IPsec, or at the application level with 

secure RTP, the real-time transport protocol (RFC 3550). However, several 

factors, including the expansion of packet size, ciphering latency, and a lack of 

QoS urgency in the cryptographic engine itself can cause an excessive amount of 

latency in the VOIP packet delivery. This leads to degraded voice quality, again 

4 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

highlighting the tradeoff between security and voice quality, and emphasizing a 

need for speed. 

VOIP is still an emerging technology, so it is difficult to develop a complete 

picture of what a mature worldwide VOIP network will one day look like. As the 

emergence of SIP has shown, new technologies and new protocol designs have 

the ability to radically change VOIP. Although there are currently many different 

architectures and protocols to choose from, eventually a true standard will 

emerge. Unless a widely used open standard emerges, solutions will be likely to 

include a number of proprietary elements, which can limit an enterprise’s future 

choices. The most widely used of the competing standards are SIP and H.323. 

Some observers believe that SIP will become dominant. Major vendors are 

investing an increasing portion of their development effort into SIP products. An 

extension of SIP, the SIP for Instant Messaging and Presence Leveraging 

Extensions (SIMPLE) standard, is being incorporated into products that support 

Instant Messaging. Until a truly dominant standard emerges, organizations 

moving to VOIP should consider gateways and other network elements that 

support both H.323 and SIP. Such a strategy helps to ensure a stable and robust 

VOIP network in the years that come, no matter which protocol prevails. 

Designing, deploying, and securely operating a VOIP network is a complex effort 

that requires careful preparation. The integration of a VOIP system into an 

already congested or overburdened network could create serious problems for the 

organization. There is no easy “one size fits all” solution to the issues discussed 

in these chapters. An organization must investigate carefully how its network is 

laid out and which solution fits its needs best. 

NIST recommendations. 

Because of the integration of voice and data in a single network, establishing a 

secure VOIP and data network is a complex process that requires greater effort 

than that required for data-only networks. In particular, start with these general 

guidelines, recognizing that practical considerations, such as cost or legal requirements, 

may require adjustments for the organization: 

1. Develop appropriate network architecture. 

• Separate voice and data on logically different networks if feasible. Different 

subnets with separate RFC 1918 address blocks should be used for voice and 

data traffic, with separate DHCP servers for each, to ease the incorporation of 

intrusion detection and VOIP firewall protection 

• At the voice gateway, which interfaces with the PSTN, disallow H.323, SIP, 

or other VOIP protocols from the data network. Use strong authentication and 

access control on the voice gateway system, as with any other critical network 

component. Strong authentication of clients towards a gateway often presents 

5 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

difficulties, particularly in key management. Here, access control mechanisms

and policy enforcement may help. 

 A mechanism to allow VOIP traffic through firewalls is required. There are a 

variety of protocol dependent and independent solutions, including application 

level gateways (ALGs) for VOIP protocols, Session Border Controllers, or 

other standards-based solutions when they mature. 

 Stateful packet filters can track the state of connections, denying packets that 

are not part of a properly originated call. (This may not be practical when 

multimedia protocol inherent security or lower layer security is applied, e.g., 

H.235 Annex D for integrity provision or TLS to protect SIP signaling.) 

 Use IPsec or Secure Shell (SSH) for all remote management and auditing 

access. If practical, avoid using remote management at all and do IP PBX 

access from a physically secure system. 

• If performance is a problem, use encryption at the router or other gateway, not 

the individual endpoints, to provide for IPsec tunneling. Since some VOIP 

endpoints are not computationally powerful enough to perform encryption, 

placing this burden at a central point ensures all VOIP traffic emanating from

the enterprise network has been encrypted. Newer IP phones are able to 

provide Advanced Encryption System (AES) encryption at reasonable cost. 

Note that Federal Information Processing Standard (FIPS) 140-2, Security 

Requirements for Cryptographic Modules, is applicable to all Federal agencies 

that use cryptographic-based security systems to protect sensitive information 

in computer and telecommunication systems (including voice systems) as 

defined in Section 5131 of the Information Technology Management Reform

Act of 1996, Public Law 104-106. 

2. Ensure that the organization has examined and can acceptably manage 

and mitigate the risks to their information, system operations, and 

continuity of essential operations when deploying VOIP systems. 

An especially challenging security environment is created when new technologies 

are deployed. Risks often are not fully understood, administrators are not yet 

experienced with the new technology, and security controls and policies must be 

updated. Therefore, agencies should carefully consider such issues as their level 

of knowledge and training in the technology, the maturity and quality of their 

security practices, controls, policies, and architectures, and their understanding of 

the associated security risks. These issues should be considered for all systems 

but are especially important with VOIP deployment for essential operations, such 

as systems designated "high" under FIPS 199, Standards for Security 

Categorization of Federal Information and Information Systems [2]. 

VOIP can provide more flexible service at lower cost, but there are significant 

tradeoffs that must be considered. VOIP systems can be expected to be more 

6 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

vulnerable than conventional telephone systems, in part because they are tied in to 

the data network, resulting in additional security weaknesses and avenues of 

attack (see Appendix A for more detailed discussion of vulnerabilities of VOIP 

and their relation to data network vulnerabilities). Confidentiality and privacy 

may be at greater risk in VOIP systems unless strong controls are implemented 

and maintained. An additional concern is the relative instability of VOIP 

technology compared with established telephony systems. Today, VOIP systems 

are still maturing and dominant standards have not emerged. This instability is 

compounded by VOIP’s reliance on packet networks as a transport medium. The 

public switched telephone network is ultra-reliable. Internet service is generally 

much less reliable, and VOIP cannot function without Internet connections, 

except in the case of large corporate or other users who may operate a private 

network. Essential telephone services, unless carefully planned, deployed, and 

maintained, will be at greater risk if based on VOIP. 

3. Special consideration should be given to E-911 emergency services 

communications, because E-911 automatic location service is not available 

with VOIP in some cases. 

Unlike traditional telephone connections, which are tied to a physical location, 

VOIP’s packet switched technology allows a particular number to be anywhere. 

This is convenient for users, because calls can be automatically forwarded to their 

locations. But the tradeoff is that this flexibility severely complicates the 

provision of E-911 service, which normally provides the caller’s location to the 

911 dispatch office. Although most VOIP vendors have workable solutions for E911 service, government regulators and vendors are still working out standards 

and procedures for 911 services in a VOIP environment. Agencies must carefully 

evaluate E-911 issues in planning for VOIP deployment. 

4. Agencies should be aware that physical controls are especially

important in a VOIP environment and deploy them accordingly. 

Unless the VOIP network is encrypted, anyone with physical access to the office 

LAN could potentially connect network monitoring tools and tap into telephone 

conversations. Although conventional telephone lines can also be monitored when 

physical access is obtained, in most offices there are many more points to 

connect with a LAN without arousing suspicion. Even if encryption is used, 

physical access to VOIP servers and gateways may allow an attacker to do traffic 

analysis (i.e., determine which parties are communicating). Agencies therefore 

should ensure that adequate physical security is in place to restrict access to VOIP 

network components. Physical securities measures, including barriers, locks, 

access control systems, and guards, are the first line of defense. Agencies must 

make sure that the proper physical countermeasures are in place to mitigate some

of the biggest risks such as insertion of sniffers or other network monitoring 

devices. Otherwise, practically speaking this means that installation of a sniffer 

could result in not just data but all voice communications being intercepted. 

7 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

5. Evaluate costs for additional power backup systems that may be 

required to ensure continued operation during power outages. 

A careful assessment must be conducted to ensure that sufficient backup power is 

available for the office VOIP switch, as well as each desktop instrument. Costs 

may include electrical power to maintain UPS battery charge, periodic 

maintenance costs for backup power generation systems, and cost of UPS battery 

replacement. If emergency/backup power is required for more than a few hours, 

electrical generators will be required. Costs for these include fuel, fuel storage 

facilities, and cost of fuel disposal at end of storage life. 

6. VOIP-ready firewalls and other appropriate protection mechanisms 

should be employed. Agencies must enable, use, and routinely test the 

security features that are included in VOIP systems. 

Because of the inherent vulnerabilities (e.g. susceptibility to packet sniffing) when 

operating telephony across a packet network, VOIP systems incorporate an array 

of security features and protocols. Organization security policy should ensure that 

these features are used. Additional measures, described in this document, should 

be added. In particular, firewalls designed for VOIP protocols are an essential 

component of a secure VOIP system. 

7. If practical, “softphone” systems, which implement VOIP using an 

ordinary PC with a headset and special software, should not be used where 

security or privacy are a concern. 

Worms, viruses, and other malicious software are extraordinarily common on PCs 

connected to the internet, and very difficult to defend against. Well-known 

vulnerabilities in web browsers make it possible for attackers to download 

malicious software without a user’s knowledge, even if the user does nothing 

more than visit a compromised web site. Malicious software attached to email 

messages can also be installed without the user’s knowledge, in some cases even 

if the user does not open the attachment. These vulnerabilities result in 

unacceptably high risks in the use of “softphones”, for most applications. In 

addition, because PCs are necessarily on the data network, using a softphone 

system conflicts with the need to separate voice and data networks to the greatest 

extent practical. 

8. If mobile units are to be integrated with the VOIP system, use products 

implementing WiFi Protected Access (WPA), rather than 802.11 Wired 

Equivalent Privacy (WEP). 

The security features of 802.11 WEP provide little or no protection because WEP 

can be cracked with publicly available software. The more recent WiFi Protected 

Access (WPA), a snapshot of the ongoing 802.11i standard, offers significant 

improvements in security, and can aid the integration of wireless technology with 

8 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

VOIP. NIST strongly recommends that the WPA (or WEP if WPA is 

unavailable) security features be used as part of an overall defense-in-depth 

strategy. Despite their weaknesses, the 802.11 security mechanisms can provide 

a degree of protection against unauthorized disclosure, unauthorized network 

access, or other active probing attacks. However, the Federal Information 

Processing Standard (FIPS) 140-2, Security Requirements for Cryptographic 

Modules, is mandatory and binding for Federal agencies that have determined that 

certain information must be protected via cryptographic means. As currently 

defined, neither WEP nor WPA meets the FIPS 140-2 standard. In these cases, it 

will be necessary to employ higher level cryptographic protocols and applications

such as secure shell (SSH), Transport Level Security (TLS) or Internet Protocol 

Security (IPsec) with FIPS 140-2 validated cryptographic modules and associated 

algorithms to protect information, regardless of whether the nonvalidated data 

link security protocols are used. 

9. Carefully review statutory requirements regarding privacy and record 

retention with competent legal advisors. 

Although legal issues regarding VOIP are beyond the scope of this document, 

readers should be aware that laws and rulings governing interception or monitoring 

of VOIP lines, and retention of call records, may be different from those for 

conventional telephone systems. Agencies should review these issues with their 

legal advisors. See Section 2.5 for more on these issues. 

9 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

1 Introduction

Voice over Internet Protocol (VOIP) refers to the transmission of speech across 

data-style networks. This form of transmission is conceptually superior to 

conventional circuit switched communication in many ways. However, a plethora 

of security issues are associated with still-evolving VOIP technology. This 

publication introduces VOIP, its security challenges, and potential 

countermeasures for VOIP vulnerabilities. 

1.1 Authority

The National Institute of Standards and Technology (NIST) developed this 

document in furtherance of its statutory responsibilities under the Federal 

Information Security Management Act (FISMA) of 2002, Public Law 107-347. 

NIST is responsible for developing standards and guidelines, including minimum 

requirements, for providing adequate information security for all agency 

operations and assets, but such standards and guidelines shall not apply to 

national security systems. This guideline is consistent with the requirements of 

the Office of Management and Budget (OMB) Circular A-130, Section 8b(3), 

“Securing Agency Information Systems,” as analyzed in A-130, Appendix IV: 

Analysis of Key Sections. Supplemental information is provided in A-130, 

Appendix III. 

This guideline has been prepared for use by Federal agencies. It may be used by 

nongovernmental organizations on a voluntary basis and is not subject to 

copyright, though attribution is desired. 

Nothing in this document should be taken to contradict standards and guidelines 

made mandatory and binding on Federal agencies by the Secretary of Commerce 

under statutory authority, nor should these guidelines be interpreted as altering or 

superseding the existing authorities of the Secretary of Commerce, Director of the 

OMB, or any other Federal official. 

1.2 Document Scope and Purpose 

The purpose of this document is to provide agencies with guidance for 

establishing secure VOIP networks. Agencies are encouraged to tailor the 

recommended guidelines and solutions to meet their specific security or business 

requirements. VOIP security considerations for the public switched telephone 

network are largely outside the scope of this document. Although legal issues 

regarding VOIP are beyond the scope of this document, readers should be aware 

that laws and rulings governing interception or monitoring of VOIP lines, and 

retention of call records, may be different from those for conventional telephone 

systems. Agencies should review these issues with their legal advisors. See 

Section 2.5 for more on this issue. 

10 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

1.3 Audience andAssumptions 

VOIP is a very large, complex, and rapidly evolving field. This document reviews 

VOIP technologies and solutions. Each section provides background information 

for the reader who is new to VOIP, but most sections also include details of 

standards and technologies that may be of interest to technical personnel only. 

The following list highlights how people with different backgrounds might use 

this document. The intended audience includes the following: 

 Managers planning to employ VOIP telephony devices in their organizations 

(chief information officers, senior managers, etc). 

 Systems engineers and architects when designing and implementing networks. 

 System administrators when administering, patching, securing or upgrading 

networks that include VOIP components. 

 Security consultants when performing security assessments to determine security 

postures of VOIP environments. 

 Researchers and analysts interested in VOIP technologies. 

This document assumes that the readers have some minimal operating system, 

networking, and security expertise. Because of the rapidly changing nature of the 

telecommunications industry and the threats and vulnerabilities to these 

technologies, readers are strongly encouraged to take advantage of other resources 

(including those listed in this document) for more current and detailed 

information. 

1.4 Document Organization 

The document is divided into five sections followed by three appendices. This 

subsection is a roadmap describing the document structure. 

 Section 1 is composed of an authority, purpose, scope, audience, assumptions, and 

document structure. 

 Section 2 provides an overview of VOIP technology. 

 Section 3 discusses performance and quality of service aspects of VOIP, and their 

effect on security options. 

 Sections 4 and 5 describe the two most commonly used protocols for VOIP: 

H.323 and Session Initiation Protocol (SIP). 

11 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

 Section 6 discusses media decomposition using the two most commonly used 

standards MGCP and Megaco/H.248. 

 Section 7 explains requirements for call establishment and address translation in a 

packet-switched telephony system, and the impact of these issues on VOIP 

security. 

 Section 8 discusses encryption technologies that can be employed in a VOIP 

network. 

 Section 9 summarizes the options available for securing VOIP systems. 

 Section 10 highlights issues to be considered in planning for VOIP. 

 

 Appendix A describes common risks, threats, and vulnerabilities of VOIP 

systems, to aid system administrators in securing their systems. 

 Appendix B covers some of the frequently asked questions regarding VOIP 

technology. 

 Appendix C provides a glossary of terms and acronyms used in this document. 

12 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

2 Overview of VOIP 

Many readers who have a good understanding of the Internet and data 

communications technology may have little background in transmitting voice or 

real-time imaging in a packet-switched environment. One of the main sources of 

confusion for those new to VOIP is the (natural) assumption that because 

digitized voice travels in packets just like other data, existing network 

architectures and tools can be used without change for voice transmission. VOIP 

adds a number of complications to existing network technology, and these 

problems are compounded by security considerations. Most of this report is 

focused on how to overcome the complications introduced by security 

requirements for VOIP. 

For several years, VOIP was a technology prospect, something on the horizon for 

the “future works” segment of telephony and networking papers. Now, however, 

telecommunications companies and other organizations have already, or are in the 

process of, moving their telephony infrastructure to their data networks. The 

VOIP solution provides a cheaper and clearer alternative to traditional PSTN 

phone lines. Although its implementation is widespread, the technology is still 

developing. It is growing rapidly throughout North America and Europe, but it is 

sometimes awkwardly implemented on most legacy networks, and often lacks 

compatibility and continuity with existing systems. Nevertheless, VOIP will 

capture a significant portion of the telephony market, given the fiscal savings and 

flexibility that it can provide. 

2.1 VOIP Equipment 

VOIP systems take a wide variety of forms. Just about any computer is capable 

of providing VOIP; Microsoft’s NetMeeting, which comes with any Windows 

platform, provides some VOIP services, as does the Apple Macintosh iChat, and 

Linux platforms have a number of VOIP applications to choose from. In general, 

though, the term Voice Over IP is associated with equipment that provides the 

ability to dial telephone numbers and communicate with parties on the other end 

of a connection who have either another VOIP system or a traditional analog 

telephone. Demand for VOIP services has resulted in a broad array of products, 

including: 

 Traditional telephone handset – Usually these products have extra features beyond 

a simple handset with dial pad. Many have a small LCD screen that may provide 

browsing, instant messaging, or a telephone directory, and which is also used in 

configuring the handset to gain access to enhanced features such as conference 

calls or call-park (automatic callback when a dialed number is no longer busy). 

Some of these units may have a “base station” design that provides the same

convenience as a conventional cordless phone. 

13 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

 Conferencing units – These provide the same type of service as conventional 

conference calling phone systems, but since communication is handled over 

the Internet, they may also allow users to coordinate data communication 

services, such as a whiteboard that displays on computer monitors at both 

ends. 

 Mobile units – Wireless VOIP units are becoming increasingly popular, 

especially since many organizations already have an installed base of 802.11 

networking equipment. Wireless VOIP products may present additional 

challenges if certain security issues are not carefully addressed. The WEP 

security features of 802.11b provide little or no protection. The more recent 

WiFi Protected Access (WPA), a snapshot of the ongoing 802.11i standard, 

offers significant improvements in security, and can aid the integration of 

wireless technology with VOIP. 

 PC or “softphone” – With a headset, software, and inexpensive connection 

service, any PC or workstation can be used as a VOIP unit, often referred to as 

a “softphone”. If practical, softphone systems should not be used where 

security or privacy are a concern. Worms, viruses, and other malicious 

software are common on PCs connected to the internet, and very difficult to 

defend against. Well known vulnerabilities in web browsers make it possible 

for attackers to download malicious software without a user’s knowledge, 

even if the user does nothing more than visit a compromised web site. 

Malicious software attached to email messages can also be installed without 

the user’s knowledge, in some cases even if the user does not open the 

attachment. These vulnerabilities result in unacceptably high risks in the use 

of “softphones”, for most applications. In addition, because PCs are 

necessarily on the data network, using a softphone system conflicts with the 

need to separate voice and data networks to the greatest extent practical. 

In addition to end-user equipment, VOIP systems include a large number of other 

components, including call processors (call managers), gateways, routers, 

firewalls, and protocols. Most of these components have counterparts used in 

data networks, but the performance demands of VOIP mean that ordinary network 

software and hardware must be supplemented with special VOIP components. 

The unique nature of VOIP services has a significant impact on security 

considerations for these networks, as will be detailed in later chapters. 

2.2 Overview ofVOIP Data Handling 

Before any voice can be sent, a call must be placed. In an ordinary phone system, 

this process involves dialing the digits of the called number, which are then 

processed by the telephone company’s system to ring the called number. With 

VOIP, the user must enter the dialed number, which can take the form of a 

number dialed on a telephone keypad or the selection of a Universal Resource 

Indicator (URI), but after that a complex series of packet exchanges must occur, 

14 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

based on a VOIP signaling protocol. The problem is that computer systems are 

addressed using their IP address, but the user enters an ordinary telephone number 

or URI to place the call. The telephone number or URI must be linked with an IP 

address to reach the called party, much as an alphabetic web address, such as 

“www.nist.gov” must be linked to the IP address of the NIST web server. A 

number of protocols are involved in determining the IP address that corresponds 

to the called party’s telephone number. This process is covered in detail in 

Chapter 6. 

Analog-digital 

converter 

Data

Compression 

RTP 

UDP packet 

Alice’s voice 

terminal 

(telephone or

computer) 

INTERNET 

Bob’s voice terminal 

(telephone or

computer) 

Figure 1. Voice Data Processing in a VOIP System. 

15 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

Figure 1 illustrates the basic flow of voice data in a VOIP system. Once the 

called party answers, voice must be transmitted by converting the voice into 

digitized form, then segmenting the voice signal into a stream of packets. The 

first step in this process is converting analog voice signals to digital, using an 

analog-digital converter. Since digitized voice requires a large number of bits, a 

compression algorithm can be used to reduce the volume of data to be transmitted. 

Next, voice samples are inserted into data packets to be carried on the Internet. 

The protocol for the voice packets is typically the Real-time Transport Protocol, 

RTP (RFC 3550). RTP packets have special header fields that hold data needed 

to correctly re-assemble the packets into a voice signal on the other end. But 

voice packets will be carried as payload by UDP protocols that are also used for 

ordinary data transmission. In other words, the RTP packets are carried as data by 

the UDP datagrams, which can then be processed by ordinary network nodes 

throughout the Internet. At the other end, the process is reversed: the packets are 

disassembled and put into the proper order, digitized voice data extracted from the 

packets and uncompressed, then the digitized voice is processed by a digital-toanalog converter to render it into analog signals for the called party’s handset 

speaker. 

2.3 Cost 

The feature of VOIP that has attracted the most attention is its cost-saving 

potential. By moving away from the public switched telephone networks, long 

distance phone calls become very inexpensive. Instead of being processed across 

conventional commercial telecommunications line configurations, voice traffic 

travels on the Internet or over private data network lines. 

VOIP is also cost effective because all of an organization’s electronic traffic 

(phone and data) is condensed onto one physical network, bypassing the need for 

separate PBX tie lines. Although there is a significant initial startup cost to such 

an enterprise, significant net savings can result from managing only one network 

and not needing to sustain a legacy telephony system in an increasingly 

digital/data centered world. Also, the network administrator’s burden may be 

lessened as they can now focus on a single network. There is no longer a need for 

several teams to manage a data network and another to mange a voice network. 

The simplicity of VOIP systems is attractive, one organization / one network; but 

as we shall see, the integration of security measures into this architecture is very 

complex. 

2.4 Speed and Quality

In theory, VOIP can provide reduced bandwidth use and quality superior to its 

predecessor, the conventional PSTN. That is, the use of high bandwidth media 

common to data communications, combined with the high quality of digitized 

voice, make VOIP a flexible alternative for speech transmission. In practice, 

however, the situation is more complicated. Routing all of an organization’s 

16 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

traffic over a single network causes congestion and sending this traffic over the 

Internet can cause a significant delay in the delivery of speech. Also, bandwidth 

usage is related to digitization of voice by codecs, circuits or software processes 

that code and decode data for transmission. That is, producing greater bandwidth 

savings may slow down encoding and transmission processes. Speed and voice 

quality improvements are being made as VOIP networks and phones are deployed 

in greater numbers, and many organizations that have recently switched to a 

VOIP scheme have noticed no significant degradation in speed or quality. 

2.5 Privacy and Legal Issues with VOIP

Although legal issues regarding VOIP are beyond the scope of this document, 

readers should be aware that laws and rulings governing interception or 

monitoring of VOIP lines may be different from those for conventional telephone 

systems. Privacy issues, including the security of call detail records (CDR) are 

addressed primarily by the Privacy Act of 1974. In addition, agencies may need 

to consider the Office of Management and Budget's "Guidance on the Privacy Act 

Implications of Call Detail Programs to Manage Employees' Use of the 

Government's Telecommunication System" (See FEDERAL REGISTER, 52 FR 

12990, April 20, 1987). Because of these guidelines, many federal agencies have 

Privacy Act System of Record notices for the telephone CDR or usage records. 

CDR data may be used to reconcile the billing of services and for possible 

detection of waste, fraud, and abuse of government resources. In addition, NARA 

General Records Schedule 12, requires a 36-month retention of telephone CDR 

records (see http://www.archives.gov/records_management/ardor/grs12.html).

VOIP systems may produce different types (and a higher volume) of CDR data 

than conventional telephone systems, so agencies must determine retention 

requirements for these records. Agencies should review any questions regarding 

privacy and statutory concerns with their legal advisors. 

2.6 VOIP SecurityIssues

With the introduction of VOIP, the need for security is compounded because now 

we must protect two invaluable assets, our data and our voice. Federal 

government agencies are required by law to protect a great deal of information, 

even if it is unclassified. Both privacy-sensitive and financial data must be 

protected, as well as other government information that is categorized as sensitive 

but unclassified. Protecting the security of conversations is thus required. In a 

conventional office telephone system, security is a more valid assumption. 

Intercepting conversations requires physical access to telephone lines or 

compromise of the office private branch exchange (PBX). Only particularly 

security-sensitive organizations bother to encrypt voice traffic over traditional 

telephone lines. The same cannot be said for Internet-based connections. For 

example, when ordering merchandise over the phone, most people will read their 

credit card number to the person on the other end. The numbers are transmitted 

without encryption to the seller. In contrast, the risk of sending unencrypted data 

17 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

across the Internet is more significant. Packets sent from a user’s home computer 

to an online retailer may pass through 15-20 systems that are not under the control 

of the user’s ISP or the retailer. Because digits are transmitted using a standard 

for transmitting digits out of band as special messages, anyone with access to 

these systems could install software that scans packets for credit card information. 

For this reason, online retailers use encryption software to protect a user’s 

information and credit card number. So it stands to reason that if we are to

transmit voice over the Internet Protocol, and specifically across the Internet, 

similar security measures must be applied. 

The current Internet architecture does not provide the same physical wire security 

as the phone lines. The key to securing VOIP is to use the security mechanisms

like those deployed in data networks (firewalls, encryption, etc.) to emulate the 

security level currently enjoyed by PSTN network users. This publication 

investigates the attacks and defenses relevant to VOIP and explores ways to 

provide appropriate levels of security for VOIP networks at reasonable cost. 

18 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

3 Quality of Service Issues 

Quality of Service (QoS) [3,4] is fundamental to the operation of a VOIP 

network. Despite all the money VOIP can save users and the network elegance it 

provides, if it cannot deliver at least the same quality of call setup and voice relay 

functionality and voice quality as a traditional telephone network, then it will 

provide little added value. The implementation of various security measures can 

degrade QoS. These complications range from delaying or blocking of call setups 

by firewalls to encryption-produced latency and delay variation (jitter). QoS 

issues are central to VOIP security. If QoS was assured, then most of the same

security measures currently implemented in today’s data networks could be used 

in VOIP networks. But because of the time-critical nature of VOIP, and its low 

tolerance for disruption and packet loss, many security measures implemented in 

traditional data networks just aren’t applicable to VOIP in their current form. The 

main QoS issues associated with VOIP that security affects are presented here: 

3.1 Latency

Latency in VOIP refers to the time it takes for a voice transmission to go from its 

source to its destination. Ideally, we would like to keep latency as low as possible 

but there are practical lower bounds on the delay of VOIP. The ITU-T 

Recommendation G.114 [5] establishes a number of time constraints on one-way 

latency. The upper bound is150 ms for one-way traffic. This corresponds to the 

current latency bound experienced in domestic calls across PSTN lines in the 

continental United States [6]. For international calls, a delay of up to 400 ms was 

deemed tolerable [7], but since most of the added time is spent routing and 

moving the data over long distances, we consider here only the domestic case and 

assume our solutions are upwards compatible in the international realm. 

VOIP calls must achieve the 150 ms bound to successfully emulate the QoS that 

today’s phones provide. This time constraint leaves very little margin for error in 

packet delivery. Furthermore, it places a genuine constraint on the amount of 

security that can be added to a VOIP network. The encoding of voice data can 

take between 1 and 30 ms [8] and voice data traveling across the North American 

continent can take upwards of 100 ms [9] although actual travel time is often 

much faster [10]. Assuming the worst case (100 ms transfer time), 20 –50 ms

remain for queuing and security implementations. A less pessimistic delay budget 

is provided by Goode [10] and reproduced in Figure 2. (Dnw = backbone 

network delay, bounded to 29 ms with a total of 121 ms from other sources.) 

19 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

Figure 2. Sample Latency Budget

Delay is not confined to the endpoints of the system. Each hop along the network 

introduces a new queuing delay and possibly a processing delay if it is a security 

checkpoint (i.e. firewall or encryption/decryption point). Also, larger packets 

tend to cause bandwidth congestion and increased latency. In light of these 

issues, VOIP tends to work best with small packets on a logically abstracted 

network to keep latency at a minimum. 

 

3.2 Jitter

Jitter refers to non-uniform packet delays. It is often caused by low bandwidth 

situations in VOIP and can be exceptionally detrimental to the overall QoS. 

Variations in delays can be more detrimental to QoS than the actual delays 

themselves [11]. Jitter can cause packets to arrive and be processed out of

sequence. RTP, the protocol used to transport voice media, is based on UDP so 

packets out of order are not reassembled at the protocol level. However, RTP 

allows applications to do the reordering using the sequence number and 

timestamp fields. The overhead in reassembling these packets is non-trivial, 

especially when dealing with the tight time constraints of VOIP. 

When jitter is high, packets arrive at their destination in spurts. This situation is 

analogous to uniform road traffic coming to a stoplight. As soon as the stoplight 

turns green (bandwidth opens up), traffic races through in a clump. The general 

prescription to control jitter at VOIP endpoints is the use of a buffer, but such a 

buffer has to release its voice packets at least every 150 ms (usually a lot sooner 

given the transport delay) so the variations in delay must be bounded. The buffer 

implementation issue is compounded by the uncertainty of whether a missing 

packet is simply delayed an anomalously long amount of time, or is actually lost. 

If jitter is particularly erratic, then the system cannot use past delay times as an 

indicator for the status of a missing packet. This leaves the system open to 

implementation specific behavior regarding such a packet. 

20 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

Jitter can also be controlled throughout the VOIP network by using routers, 

firewalls, and other network elements that support QoS. These elements process 

and pass along time urgent traffic like VOIP packets sooner than less urgent data 

packets. However, not all network components were designed with QoS in mind. 

An example of a network element that does not implement this QoS demand is a 

crypto-engine that ignores Type of Service (ToS) bits in an IP header and other 

indicators of packet urgency (see Section 8.6). Another method for reducing 

delay variation is to pattern network traffic to diminish jitter by making as 

efficient use of the bandwidth as possible. This constraint is at odds with some

security measures in VOIP. Chief among these is IPsec, whose processing 

requirements may increase latency, thus limiting effective bandwidth and

contributing to jitter. Effective bandwidth is compromised when packets are 

expanded with new headers. In normal IP traffic, this problem is negligible since 

the change in the size of the packet is very small compared with the packet size. 

Because VOIP uses very small packets, even a minimal increase is important 

because the increase accrues across all the packets, and VOIP sends a very high 

volume of these small packets. 

The window of delivery for a VOIP packet is very small, so it follows that the 

acceptable variation in packet delay is even smaller. Thus, although we are 

concerned with security, the utmost care must be given to assuring that delays in 

packet deliveries caused by security devices are kept uniform throughout the 

traffic stream. Implementing devices that support QoS and improving the 

efficiency of bandwidth with header compression allows for more uniform packet 

delay in a secured VOIP network. 

3.3 Packet Loss 

VOIP is exceptionally intolerant of packet loss. Packet loss can result from

excess latency, where a group of packets arrives late and must be discarded in 

favor of newer ones. It can also be the result of jitter, that is, when a packet 

arrives after its surrounding packets have been flushed from the buffer, making 

the received packet useless. VOIP-specific packet loss issues exist in addition to 

the packet loss issues already associated with data networks; these are the cases 

where a packet is not delivered at all. Compounding the packet loss problem is 

VOIP’s reliance on RTP, which uses the unreliable UDP for transport, and thus 

does not guarantee packet delivery. However, the time constraints do not allow 

for a reliable protocol such as TCP to be used to deliver media. By the time a 

packet could be reported missing, retransmitted, and received, the time constraints 

for QoS would be well exceeded. The good news is that VOIP packets are very 

small, containing a payload of only 10-50 bytes [8], which is approximately 12.5-

62.5 ms, with most implementations tending toward the shorter range. The loss of 

such a minuscule amount of speech is not discernable or at least not worthy of 

complaint for a human VOIP user. The bad news is these packets are usually not 

lost in isolation. Bandwidth congestion and other such causes of packet loss tend 

to affect all the packets being delivered around the same time. So although the 

21 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

loss of one packet is fairly inconsequential, probabilistically the loss of one packet 

means the loss of several packets, which severely degrades the quality of service 

in a VOIP network. 

In a comparison of VOIP quality versus traditional circuit switched networks, 

Sinden [12] reported data from a Telecommunications Industry Association (TIA) 

study that showed even a fairly small percentage of lost packets could push VOIP 

network QoS below the level users have come to expect on their traditional phone 

lines. Each codec the TIA studied experienced a steep downturn in user 

satisfaction when latency crossed the 150 ms point. However, even with less than 

150 ms of latency, a packet loss of 5% caused VOIP traffic encoded with G.711 

(an international standard for encoding telephone audio on a 64 kbps stream) to 

drop below the QoS levels of the PSTN, even with a packet loss concealment 

scheme. Similarly, losses of 1 and 2 percent, respectively, were enough to place 

quality in VOIP networks encoded with G.723.1 (for very low bit rate speech 

compression) and G.729A (for voice compression on an 8kbps stream) below this 

threshold. At losses of 3 and 4 percent, respectively, the performance of these 

networks resulted in a majority of dissatisfied users. These results corroborated 

the findings of a 1998 study at UCal-Berkeley that determined “tolerable loss 

rates are within 1-3% and the quality becomes intolerable when more than 3% of 

the voice packets are lost” [9]. Both studies found that greater payload 

compression rates resulted in a higher sensitivity to packet loss. On the bright 

side, the implementation of forward error correction [9] and packet loss 

concealment schemes produced a VOIP network that was less sensitive to packet 

loss. The percentages presented in both studies did not take into account varying 

packet sizes and several other properties that can affect the relationship between 

packet loss and QoS. 

Despite the infeasibility of using a guaranteed delivery protocol such as TCP, 

there are some remedies for the packet loss problem. One cannot guarantee all 

packets are delivered, but if bandwidth is available, sending redundant 

information can probabilistically annul the chance of loss. Such bandwidth is not 

always accessible and the redundant information will have to be processed, 

introducing even more latency to the system and ironically, possibly producing 

even greater packet loss. Newer codecs such as internet Low Bit-rate Codec 

(iLBC) are also being developed that offer roughly the voice quality and 

computational complexity of G.729A, while providing increased tolerance to 

packet loss.

3.4 Bandwidth & Effective Bandwidth 

In any network, the obvious first concern is whether the network is available for 

use. Since a network can be broken down into nodes and links between nodes 

where traffic flows, the quest for an available network boils down to the 

availability of each node, and the availability of each path between the nodes. 

Later on, we will consider the nodes themselves, in cases where firewalls, CPUs, 

22 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

or other endpoints are unavailable, but for now we will concentrate on the 

availability of the edges: the bandwidth of the VOIP system. 

As in data networks, bandwidth congestion can cause packet loss and a host of 

other QoS problems. Thus, proper bandwidth reservation and allocation is 

essential to VOIP quality. One of the great attractions of VOIP, data and voice 

sharing the same wires, is also a potential headache for implementers who must 

allocate the necessary bandwidth for both networks in a system normally designed 

for one. Congestion of the network causes packets to be queued, which in turn 

contributes to the latency of the VOIP system. Low bandwidth can also 

contribute to non-uniform delays (jitter), since packets will be delivered in spurts 

when a window of opportunity opens up in the traffic. 

Because of these issues, VOIP network infrastructures must provide the highest 

amount of bandwidth possible. On a LAN, this means having modern switches 

running at 100M bit/sec and other architectural upgrades that will alleviate 

bottlenecks within the LAN. Percy and Hommer [13] suggest that if network 

latencies are kept below 100 milliseconds, maximum jitter never more than 40 

milliseconds, then packet loss should not occur. With these properties assured, 

one can calculate the necessary bandwidth for a VOIP system on the LAN in a 

worst case scenario using statistics associated with the worst-case bandwidth 

congesting codec [13]. This is fine when dealing simply with calls across the 

LAN, but the use of a WAN complicates matters. Bandwidth usage varies 

significantly across a WAN, so a much more complex methodology is needed to 

estimate required bandwidth usage. Chuah [9] provides an analysis of the 

aggregate bandwidth needed in terms of the amount of traffic and its rate of flow. 

Methods for reducing the bandwidth usage of VOIP include RTP header 

compression and Voice Activity Detection (VAD). RTP compression condenses 

the media stream traffic so less bandwidth is used. However, an inefficient 

compression scheme can cause latency or voice degradation, causing an overall 

downturn in QoS. VAD prevents the transmission of empty voice packets (i.e. 

when a user is not speaking, their device does not simply send out white noise). 

However, by definition VAD will contribute to jitter in the system by causing 

irregular packet generation. 

The bandwidth requirements put forth by Chuah are designed for a basic VOIP 

system. Adding security constraints significantly increases the bandwidth usage, 

causing more latency and jitter, thereby degrading the overall QoS of the network. 

In addition, these requirements do not explicitly take into account the 

heterogeneous data flow over the network. Since voice and data streams are 

sharing the same finite bandwidth, and data streams tend to contain much larger 

packets than VOIP, significant amounts of data can congest the network and 

prevent voice traffic from reaching its destination in a timely fashion. For this 

reason, most new hardware devices deployed on networks support QoS for VOIP. 

These devices, such as routers and firewalls, make use of the IP protocol’s Type 

23 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

of Service (ToS) bits to send VOIP traffic through before less time urgent data 

traffic. VOIP phones often also include QoS features [13]. 

Not only is the available bandwidth of the system affected by the introduction of 

security measures, but in addition the effective bandwidth of the VOIP system is 

significantly depreciated. Effective bandwidth is defined by Barbieri et al. [8] as 

“the percentage of bandwidth carrying actual data with regard to the total 

bandwidth used.” The introduction of IPsec or other forms of encryption results 

in a much larger header to payload ratio for each packet, and this reduces the 

effective bandwidth as the same number of packets (but larger sized) are used to 

transport the same amount of data. The consequences of this reduction include 

decreased throughput and increased latency. 

3.5 The Needfor Speed 

The key to conquering QoS issues like latency and bandwidth congestion is 

speed. By definition, faster throughput means reduced latency and 

probabilistically reduces the chances of severe bandwidth congestion. Thus every 

facet of network traversal must be completed quickly in VOIP. The latency often 

associated with tasks in data networks will not be tolerated. Chief among these 

latency producers that must improve performance are firewall/NAT traversal and 

traffic encryption/decryption. Traditionally, these are two of the most effective 

ways for administrators to secure their networks. However, they are also two of 

the greatest contributors to network congestion and throughput delay. Inserting 

traditional firewall and encryption products into a VOIP network is not feasible, 

particularly when VOIP is integrated into existing data networks. Instead, these 

data-network solutions must be adapted to support security in the new fast paced 

world of VOIP. The next several chapters explore the resolution of the conflict 

between the speed demands of QoS and the slowdown associated with these 

traditional security measures. 

3.6 Power Failureand Backup Systems

Conventional telephones operate on 48 volts supplied by the telephone line itself. 

This is why home telephones continue to work even during a power failure. Most 

offices use PBX systems with their conventional telephones, and PBXs require 

backup power systems so that they continue to operate during a power failure. 

These backup systems will continue to be required with VOIP, and in many cases 

will need to be expanded. An organization that provides uninterruptible power 

systems for its data network and desktop computers may have much of the power 

infrastructure needed to continue communication functions during power outages, 

but a careful assessment must be conducted to ensure that sufficient backup power 

is available for the office VOIP switch, as well as each desktop instrument. 

Costs may include electrical power to maintain UPS battery charge, periodic 

maintenance costs for backup power generation systems, and cost of UPS battery 

replacement. If emergency/backup power is required for more than a few hours, 

24 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

electrical generators will be required. Costs for these include fuel, fuel storage 

facilities, and cost of fuel disposal at end of storage life. 

3.7 Quality of Service Implications for Security

The strict performance requirements of VOIP have significant implications for 

security, particularly denial of service (DoS) issues [14]. VOIP-specific attacks 

(i.e., floods of specially crafted SIP messages) may result in DoS for many VOIPaware devices. For example, SIP phone endpoints may freeze and crash when 

attempting to process a high rate of packet traffic SIP proxy servers also may 

experience failure and intermittent log discrepancies with a VOIP-specific

signaling attack of under 1Mb/sec. In general, the packet rate of the attack may 

have more impact than the bandwidth; i.e., a high packet rate may result in a 

denial of service even if the bandwidth consumed is low. 

25 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

4 H.323 

H.323 is the ITU specification for audio and video communication across 

packetized networks. H.323 is actually an umbrella standard, encompassing 

several other protocols, including H.225, H.245, and others. It acts as a wrapper 

for a suite of media control recommendations by the ITU. Each of these protocols 

has a specific role in the call setup process, and all but one are made to dynamic 

ports. Figure 3 shows the H.323 architecture and Figure 4 provides an overview 

of the H.323 call setup process. 

4.1 H.323 Architecture 

An H.323 network is made up of several endpoints (terminals), a gateway, and 

possibly a gatekeeper, Multipoint control unit, and Back End Service. The 

gatekeeper is often one of the main components in H.323 systems. It provides 

address resolution and bandwidth control. The gateway serves as a bridge

between the H.323 network and the outside world of (possibly) non-H.323 

devices. This includes SIP networks and traditional PSTN networks. This 

brokering can add to delays in VOIP, and hence there has been a movement 

towards the consolidation of at least the two major VOIP protocols [15]. A 

Multipoint Control Unit is an optional element that facilitates multipoint 

conferencing and other communications between more than two endpoints. 

Gatekeepers are an optional but widely used component of a VOIP network [16]. 

If a gatekeeper is present, a Back End Service (BES) may exist to maintain data 

about endpoints, including their permissions, services, and configuration [17]. 

Figure 3. H.323 Architecture 

26 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

Bob TCP connection established Alice 

Call SETUP 

ALERTING Q.931 

CONNECT H.245 address/port specified 

TCP connection established 

H.245 messages 

Open logical channel (RTCP addr/port specified) 

H.245 Open logical channel ACK(RTCP & RTP addr/port)

Open logical channel (RTCP addr/port specified) 

Open logical channel ACK(RTCP & RTP addr/port)

RTP Stream

RTP Stream

RTCP Instructions 

RTP 

Figure 4. H.323 Call Setup Process 

Generally, there are different types of H.323 calls defined in the H.323 standard: 

− Gatekeeper routed call with gatekeeper routed H.245 signaling 

− Gatekeeper routed call with direct H.245 signaling 

− Direct routed call with gatekeeper 

− Direct routed call without gatekeeper

An H.323 VOIP session is initiated (depending on the call model used) by either a 

TCP or a UDP (if RAS is the starting point) connection with an H.225 signal. In 

the case of UDP this signal contains the Registration Admission Status (RAS) 

protocol that negotiates with the gatekeeper and obtains the address of the 

endpoint it is attempting to contact. Then a “Q.931-like” protocol (still within the 

realm of H.225) is used to establish the call itself and negotiate the addressing 

information for the H.245 signal. (This is done via TCP; Q.931 actually 

encapsulates the H.225 Call Signaling messages.) This “setup next” procedure is 

27 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

common throughout the H.323 progression where one protocol negotiates the 

configuration of the next protocol used. In this case, it is necessary because 

H.245 has no standard port [10]. While H.225 simply negotiates the 

establishment of a connection, H.245 establishes the channels that will actually be 

used for media transfer. Once again, this is done over TCP. In a time-urgent 

situation, the H.245 message can be embedded within the H.225 message (H.245 

tunneling), but the speed of a call setup is usually a QoS issue that vendors and 

customers are willing to concede for better call quality. H.323 also offers Fast 

Connect. Here, a call may be setup using one roundtrip. The SETUP and the 

CONNECT messages piggyback the necessary H.245 signaling elements. 

H.245 must establish several properties of the VOIP call. These include the audio 

codecs that will be used and the logical channels for the transportation of media. 

The “OpenLogicalChannel” signal also brokers the RTP and RTCP ports. 

Overall, 4 connections must be established because the logical channels (RTP and 

RTCP) are only one direction. Each one-way pair must also be on adjacent ports 

as well. After H.245 has established all the properties of the VOIP call and the 

logical channels, the call may begin. 

The preceding described the complex VOIP setup process based on H.323.. The 

H.323 suite has different protocols associated with more complex forms of 

communication including H.332 (large conferences), H.450.1, H.450.2, and 

H.450.3 (supplementary services), H.235 (security), and H.246 (interoperability 

with circuit switched services) [18]. Authentication may also be performed at

each point in the call setup process using symmetric keys or some prior shared 

secret [19]. The use of these extra protocols and/or security measures adds to the 

complexity of the H.323 setup process. We shall see that this complexity is 

paramount in the incompatibility of H.323 with firewalls and NATs. These issues 

are discussed at length in the next section. 

4.2 H.235 Security Profiles 

With the establishment of the H.235 version 2 standard in November 2000 the 

ITU-T took a step towards interoperability by defining different security profiles. 

This was necessary because the standard itself does not mandate particular 

features. The defined profiles provide different levels of security and describe a 

subset of possible security mechanisms offered by H.235. They comprise different 

options for the protection of communications, e.g., by using different options of 

H.235, which results in different implementation impact. The following 

subsections provide a short overview of the profiles provided by different 

organizations. 

28 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

4.2.1 H.235v2 

H.235v2 is the followup version of H.235 that was approved in November 2000. 

Besides enhancements such as the support of elliptic curve cryptography and the 

support for the Advanced Encryption System (AES) standard, several security 

profiles are defined to support product interoperability. These profiles are defined 

in annexes to H.235v2 as follows: 

– Annex D – Shared secrets and keyed hashes 

– Annex E – Digital signatures on every message 

– Annex F – Digital signatures and shared secret establishment on first 

handshake, afterwards keyed hash usage 

 

4.2.1.1 H.235v2 Annex D – Baseline Security Profile 

The Baseline Security Profile relies on symmetric techniques. Shared secrets are 

used to provide authentication and/or message integrity. The supported scenarios 

for this profile are endpoint to gatekeeper, gatekeeper to gatekeeper, and endpoint 

to endpoint. For the profile the gatekeeper-routed signaling (hop-by-hop security) 

is favored. Using it for the direct call model is generally possible but limited due 

to the fact that a shared secret has to be established between the parties that want 

to communicate before the actual communication takes place. This might be 

possible in smaller environments but may lead to a huge administrative effort in 

larger environments. 

This profile supports secure fast connect and H.245 tunneling, and may be

combined with the Voice Encryption Option described in section 4.2.1.3. Note: 

that this profile is easy to implement but it is not really scalable for “global” IP 

telephony due to the restricted key management. 

Security Services Call Functions 

RAS H.225.0 H.245 RTP 

Authentication Shared Secret 

(Password), 

HMAC-SHA1-96 

Shared Secret 

(Password), 

HMAC-SHA1-96 

Shared Secret 

(Password), 

HMAC-SHA1-96 

Access Control 

Non-Repudiation 

Confidentiality 

Integrity Shared Secret 

(Password), 

HMAC-SHA1-96 

Shared Secret 

(Password), 

HMAC-SHA1-96 

Shared Secret 

(Password), 

HMAC-SHA1-96 

Key Management Subscription-based 

password assignment 

Subscription-based 

password assignment 

Table 1: H235v2 Annex D - Baseline Security Profile 

29 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

4.2.1.2 H.235v2 Annex E – Signature Security Profile 

The Signature Security Profile relies on asymmetric techniques. Certificates and 

digital signatures are used to provide authentication and message integrity. The 

signature security profile mandates the gatekeeper-routed model. Other call 

models are for further study. Since this profile relies on a public key 

infrastructure rather than on pre-established shared secrets it scales for larger, 

global environments. In addition to the Baseline Security Profile it provides nonrepudiation. 

Security Services Call Functions 

RAS H.225.0 H.245 RTP 

Authentication SHA1/ MD5, 

digital signature 

SHA1/MD5, 

digital signature 

SHA1/MD5, 

digital signature 

Access Control 

Non-Repudiation SHA1/ MD5, 

digital signature 

SHA1/ MD5, 

digital signature 

SHA1/ MD5, 

digital signature 

Confidentiality 

Integrity SHA1/ MD5, 

digital signature 

SHA1/ MD5, 

digital signature 

SHA1/ MD5, 

digital signature 

Key Management certificate allocation certificate allocation

Table 2: H235v2 Annex E – Signature Security Profile 

This profile supports secure fast connect and H.245 tunneling and may be 

combined with the Voice Encryption Option described in section 4.2.1.3. Note: 

This protocol may have a critical impact on overall performance. This is due to 

the use of digital signatures for every message, requiring signature generation and 

verification on the sender’s and the receiver’s side. The Hybrid Security Profile 

described in section 4.2.1.4 provides an alternative to the Signature Security 

Profile. 

4.2.1.3 H.235v2 Annex D - Voice Encryption Option 

The voice encryption option offers confidentiality for the voice media stream data 

and may be combined with the baseline or the signature security profile. 

Security Services Call Functions 

RAS H.225.0 H.245 RTP 

Authentication 

Access Control 

Non-Repudiation 

Confidentiality 56-bit DES or 56-

RC2®/ 168-bit 

30 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

Security Services Call Functions 

RAS H.225.0 H.245 RTP 

Triple-DES, AES 

Integrity 

Key Management Authenticated DiffieHellman key 

agreement 

Integrated H.235 session key 

management (key distribution, key

update); certificate requests 

Table 3: H235v2 - Voice Encryption Option 

The voice encryption option describes the master key exchange during H.225.0 

call signaling and the generation and distribution of media stream keys during 

H.245 call control. The encryption algorithms are to be used in CBC mode. New 

is the support of the AES. AES and TDEA may also be used in EOFB mode. 

The following security mechanisms are described within the voice encryption 

security profile: 

• Encryption of RTP packets with an assortment of algorithms and modes to be 

taken; 

• Key management with key and security capability exchange; 

• Key update mechanism and synchronization. 

The following issues are not covered by this profile: 

• Encryption and key management for RTCP; 

• Authentication and integrity for RTP and RTCP (a lightweight authentication 

and integrity could be provided by media anti-spamming). 

To counter denial of service and flooding attacks on discovered RTP/UDP ports, 

the H.235 standard defines the media anti-spamming procedure, which provides 

lightweight RTP packet authentication and integrity on selected fields through a 

computed message authentication code (MAC). The algorithms used are tripleDES-MAC or the cryptographic one-way function SHA1. Media anti-spamming 

uses the padding mechanism of RTP. For this feature, no special security profile 

was specified in H.235 like the voice encryption security profile for the RTP 

encryption, but media anti-spamming may be used in combination with media 

encryption. 

4.2.1.4 H.235v2 Annex F – Hybrid Security Profile 

The Hybrid Security Profile relies on asymmetric and symmetric techniques. It 

can be seen as a combination of the Baseline and the Signature Security Profile. 

Certificates and digital signatures are used to provide authentication and message 

integrity (as in the Signature Security Profile) for the first handshake between two 

entities. During this handshake a shared secret is established that will be used 

further on in the same way described for the Baseline Security Profile. The hybrid 

security profile mandates the gatekeeper-routed model. Other call models are 

open for further study. 

31 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

Since this profile relies on a public key infrastructure rather than on preestablished shared secrets it scales for larger, global environments. 

This profile supports secure fast connect and H.245 tunneling and may be 

combined with the Voice Encryption Option described in section 4.2.1.3. 

Note: This profile provides high security without relying on pre-established 

shared secrets. Due to the key management using digital signatures it is scalable 

for “global” IP telephony. Moreover, it does not suffer from the same

performance requirements as the Signature Security Profile described in section 

4.2.1.2. 

Security Services Call Functions 

RAS H.225.0 H.245 RTP 

RSA digital 

signature,

(SHA1) 

RSA digital 

signature,

(SHA1) 

RSA digital 

signature,

(SHA1) 

Authentication

HMAC-SHA1-96 HMAC-SHA1-96 HMAC-SHA1-96

Access Control 

Non-Repudiation Only for first 

handshake send 

between two entities 

Only for first 

handshake send 

between two entities 

Confidentiality 

RSA digital 

signature,

(SHA1) 

RSA digital 

signature,

(SHA1) 

RSA digital 

signature,

(SHA1) 

Integrity 

HMAC-SHA1-96 HMAC-SHA1-96 HMAC-SHA1-96

 Key certificate allocation certificate allocation

Management authenticated DiffieHellman key 

agreement 

authenticated DiffieHellman key 

agreement 

Table 4: H235v2 Annex F – Hybrid Security Profile 

4.2.2 H.235v3 

Version 3 of H.235 supersedes H.235 version 2 featuring a procedure for 

encrypted DTMF (touch tone) signals, object identifiers for the AES encryption 

algorithm for media payload encryption, and the Enhanced Outer FeedBack 

(EOFB) stream-cipher encryption mode for encryption of media streams. 

Moreover, an authentication-only option in Annex D for smooth NAT/firewall 

traversal is introduced as well as better security support for direct-routed calls in a 

32 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

new Annex I. Error reporting is also improved. Annex G is also discussed to 

support H.235v3. Annex G describes a profile to support SRTP. 

4.2.2.1 H.235 Annex D – Baseline Security Profile Enhancements 

Using this profile, either message authentication and integrity is achieved by 

calculating an integrity check value over the complete message, or authentication 

only by computing an integrity check over a special part of the message. The 

latter option is useful in environments where NAT and Firewalls are applied. The 

version used is distinguished by an identifier. 

Security Services Call Functions 

RAS H.225.0 H.245 RTP 

Authentication Shared Secret 

(Password), 

HMAC-SHA1-96 

Shared Secret 

(Password), 

HMAC-SHA1-96 

Shared Secret 

(Password), 

HMAC-SHA1-96 

Access Control 

Non-Repudiation 

Confidentiality 

Integrity 

(optional) 

Shared Secret (Password), 

HMAC-SHA1-96 

Shared Secret (Password), 

HMAC-SHA1-96 

Shared Secret (Password), 

HMAC-SHA1-96 

Key Management Subscription-based 

password assignment 

Subscription-based 

password assignment 

Table 5: H235v3 Annex D - Baseline Security Profile 

Table 5 shows the updated version of the baseline security profile, where the 

integrity protection of the signaling data is marked as optional. 

4.2.2.2 Draft H.235v3 Annex G – SRTP & MIKEY usage 

Annex G discusses the incorporation of key management supporting the Secure 

Real-time Transport Protocol (SRTP). SRTP provides confidentiality, message 

authentication and replay protection to the RTP/RTCP traffic. The RTP standard 

provides the flexibility to adapt to application specific requirements with the 

possibility to define profiles in companion documents. SRTP is defined as such a 

profile of the RTP protocol and is currently stardards-track RFC 3711. SRTP 

may be used within multimedia sessions to ensure a secure media data exchange. 

It can be used with several session control protocols, e.g., with H.323 or SIP. 

33 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

SRTP does not define key management by itself. It rather uses a set of negotiated 

parameters from which session keys for encryption, authentication and integrity 

protection are derived. The key management is not fixed. Within the IETF, the 

MSEC working group discusses key management solutions to be used beyond 

other protocols with SRTP. The preferred solution here is Multimedia Internet 

Keying (MIKEY) [RFC 3830] which is also part of the group key management 

architecture (GKMArch). 

MIKEY describes a key management scheme that addresses real-time multimedia 

scenarios (e.g. SIP calls and RTSP sessions, streaming, unicast, groups, 

multicast). The focus lies on the setup of a security association for secure 

multimedia sessions including key management and update, security policy data,

etc., such that requirements in a heterogeneous environment are fulfilled. MIKEY 

also supports the negotiation of single and multiple crypto sessions. This is

especially useful for the case where the key management is applied to SRTP, 

since here RTP and RTCP may to be secured independently. Deployment

scenarios for MIKEY comprise peer-to-peer, simple one-to-many, and small-size 

interactive group scenarios. 

MIKEY supports the negotiation of cryptographic keys and security parameters 

(SP) for one or more security protocols. This results in the concept of crypto 

session bundles, which describe a collection of crypto sessions that may have a 

common Traffic Encryption Key (TEK), Generation Key (TGK), and session 

security parameters. 

MIKEY defines three options for the user authentication and negotiation of the

master keys all as 2 way-handshakes. They are: 

− Symmetric key distribution (pre-shared keys, MAC for integrity protection) 

− Asymmetric key distribution 

− Diffie Hellman key agreement protected by digital signatures

A fourth version exists, which is not part of MIKEY itself. It is specified as an

extension to MIKEY and describes the Diffie-Hellman key agreement protected

by symmetric pre-shared keys. 

The default and mandatory key transport encryption is AES in counter mode. 

MIKEY uses a 160-bit authentication tag, generated by HMAC with SHA-1 as 

the mandatory algorithm as described in RFC 2104. Also mandatory, when 

asymmetric mechanisms are used, is the support of X.509v3 certificates for public 

key encryption and digital signatures. 

Annex G discusses the use of MIKEY to integrate key management suitable for 

SRTP in three profiles: 

− Profile 1 using symmetric techniques to protect the key management data in 

gatekeeper routed scenarios; 

− Profile 2 using asymmetric techniques to protect the key management data 

in scenarios with a single gatekeeper instance; 

34 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

− Profile 3 describes Profile 2 for multiple intermediate gatekeepers. 

The basic concept of all profiles is the protected transmission of the key 

management data as self-contained container. 

 

4.2.2.3 Draft H.235v3 Annex H – RAS Key Management 

The basic idea formulated in H.235 Annex H is key management negotiation 

during the RAS gatekeeper discovery phase. During gatekeeper discovery a 

shared secret is established between the endpoint and the gatekeeper. The 

negotiation of the shared secret may be protected using PINs or passwords during 

the initial phase of the protocol. 

The draft references two protocols for Encrypted Key Exchange using a shared 

secret to "obscure" a Diffie-Hellman key exchange. The first one is the Encrypted 

Key Exchange (EKE), where the shared secret is used to encrypt the DiffieHellman public keys under a symmetric algorithm. The second one is the Simple 

Password-authenticated Exponential Key Exchange (SPEKE) method [20], where 

the shared secret builds a generator for the Diffie-Hellman group. The usage of 

these protocols leads to a strong Diffie-Hellman key exchange with use of the 

shared secret. A potential disadvantage of these protocols is that they are typically 

subject to patent protection. 

The draft discusses the use of the PIN or password for the protection of the 

exchange of the public parameter of public key system (Diffie Hellman, elliptic 

curves) by encryption using a symmetric algorithm in CBC mode. To be more 

specific, the password or PIN is used to derive the initialization vectors for the 

encryption algorithms. The negotiated keys and algorithms may then be applied 

later on to protect the further RAS and call signaling phase. 

One option to protect the call signaling phase is TLS, which is discussed further 

in the draft Annex H. Here, the RAS negotiation replaces the initial TLS 

handshake protocol. This is obviously only useful if the call signaling is 

gatekeeper routed. The approach is especially useful for inter-gatekeeper 

authentication and signaling using the LRQ/LCF exchange. In this case, there is 

no third RAS message by which the calling gatekeeper can authenticate itself to 

the called gatekeeper using the negotiated key material, but the caller can be 

implicitly authenticated by its ability to establish the call signaling channel with 

the correct TLS session parameters. TLS can then be deployed without the costly 

handshake phase using only the recode layer of TLS together with the negotiated 

key material and algorithms from the RAS phase. 

4.2.2.4 H.235v3 Annex I – H.235 Annex D for Direct Routed Scenarios 

Both Annex D and Annex F are to be used in gatekeeper routed environments. 

Annex I of H.235 enhances the Baseline Security Profile (Annex D, section 

4.2.1.1) as well as the Hybrid Security Profile (Annex F, section 4.2.1.3) with the 

option to be applied in an environment were direct routed calls (endpoint to 

35 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

endpoint) are performed using the gatekeeper for address resolution. Since 

endpoints do not possess a shared secret from scratch, a Kerberos-like approach is 

taken to establish a shared secret between the communicating endpoints. This is 

done using the admission phase from the calling endpoint and the call signaling 

between the calling and the called endpoint. The gatekeeper serves in this 

scenario also as the key distribution center (KDC), issuing two “tickets” (tokens), 

one containing the key material secured with the caller’s encryption key and the 

other one secured with the called party encryption key. The encryption keys are 

derived from the shared secret between the caller and the gatekeeper using a 

pseudo random function (PRF), which is also defined by H.235 Annex I. The PRF 

is basically the same as used in TLS 

The gatekeeper also generates a session key, which is applicable for the 

communication between the two endpoints involved in the call, and encrypts this 

key material using the previously derived encryption keys. The encrypted session 

keys are then transmitted back to the caller. The caller uses the encrypted session 

key destined to him, the other one is sent to the called party as part of the SETUP 

message. 

The messages exchanged between the gatekeeper and the calling endpoint 

carrying the tickets are secured with either the H.235 Annex D (section 4.2.1.1) or 

with H.235 Annex F (4.2.1.4). The shared secret established via the “ticket” 

(token) exchange between caller and callee may be used in subsequent direct 

messages to provide an integrity protection according to H.235 Annex D. 

4.2.3 H.323 Annex J 

H.323 Annex J describes security for simple endpoint types, which are defined by 

H.323 Annex F. This profile relies on the Baseline Security Profile described in 

section 4.2.1.1. 

4.2.4 H.323 Security Issues 

Firewalls pose particularly difficult problems for VOIP networks using H.323. 

With the exception of the “Q.931-like” H.225, all H.323 traffic is routed through 

dynamic ports. For H.323 Fast Start and H.245 tunneling just one channel (H.225 

Call Signaling) is used. Usually the call signaling is performed via port 1720. If 

additionally H.225 RAS communication is done with the gatekeeper (UDP), this 

is done via port 1719. That is, each successive channel in the protocol is routed 

through a port dynamically determined by its predecessor. This ad-hoc method of 

securing channels does not lend itself well to a static firewall configuration. This 

is particularly true in the case of stateless firewalls that cannot comprehend H.323 

traffic. These simple packet filters cannot correlate UDP transmissions and 

replies. This necessitates punching holes in the firewall to allow H.323 traffic to 

traverse the security bridge on any of the ephemeral ports it might use. This 

practice would introduce serious security weaknesses because such an 

36 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

implementation would need to leave 10,000 UDP ports and several H.323 specific 

TCP ports wide open [sample configuration provided in 1]. There is thus a need 

for a stateful firewall that understands VOIP, specifically H.323. Such a firewall 

can read H.323 messages and dynamically open the correct ports for each channel 

as the protocol moves through its call setup process. Such a firewall must be part 

of a security architecture especially in scenarios where protocol–provided security 

measures are applied, e.g. message integrity. Barring this, some kind of proxy 

server or middlebox would have to be used. Several solutions to this problem are 

presented in chapter 6. 

Even with a VOIP-aware firewall, parsing H.323 traffic is not a trivial matter. 

H.323 traffic is encoded in a binary format based on ASN.1. ASN.1 does not use 

fixed offsets for address information, and different instances of an application 

may negotiate different options, resulting in different byte offsets for the same

information [21]. This level of complexity does not allow for simple parsing tools 

or uncomplicated Perl scripts to decode the traffic; in fact special code generators 

are needed [18]. Such technology is not available on traditional packet filtering 

firewalls or even simple stateful firewalls. Although this analysis can be done 

using modern VOIP-aware gateways, the complex parsing necessary to discern 

the contents of the ASN.1 encoded packets introduces further latency into a 

speed-sensitive system that is already saturated with delays. 

NAT is also particularly troublesome for VOIP systems using the H.323 call setup 

protocol. NAT complicates H.323 communications because the internal IP 

address and port specified in the H.323 headers and messages themselves are not 

the actual address/port numbers used externally by a remote terminal. This 

disrupts the “setup next” procedure used by each protocol within the H.323 suite 

(e.g., H.225 setting up H.245). Not only does the firewall have to comprehend 

this, but it is essential that the VOIP application receiving these H.323 

communications receives the correct translated address/port numbers. Thus, if 

H.323 is to traverse a NAT gateway, the NAT device must be able to reconfigure 

the addresses in the control stream. So with NAT, not only does H.323 traffic 

need to be read, it must also be modified so that the correct address/port numbers 

are sent to each of the endpoints. 

4.3 EncryptionIssues and Performance 

Delay in a VOIP system can be added by codecs and by addtional processing such 

as encryption. Codecs add delay in coding and compressing speech data. 

Processing time increases with the degree of compression, because larger blocks 

of speech data are needed to produce higher degrees of compression. 

Encryption serves two purposes for VOIP: privacy protection, by encrypting 

voice data, and message authentication, which protects the origin and integrity of 

voice packets. Encryption may be done using either a stream or block cipher. If 

a stream cipher is used, very little delay is introduced if the key stream can be 

produced before or at least as fast as voice data arrives. In this case there will be 

37 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

only one bit of delay as the cipher stream is applied. Block ciphers may require 

one block of delay, which will vary with the method used, but still require 

relatively little overhead.

More significant delays are introduced by computing HMAC hash valures for 

authentication. HMAC is used with secret key hash functions, such as MD5 or 

SHA-1. HMAC-MD5 produces a 128-bit message authentication code (MAC), 

while HMAC-SHA-1 will produce a 160-bit MAC. Because the HMAC 

operation must wait for a full block of data to arrive before processing, these 

operations can produce significant latency delays. On arrival, the reverse

operations must be applied, introducing further performance delays. 

In most applications, authentication and integrity are equally, or more, important 

than encryption, but with voice processing for human speakers, some

authentication is normally built-in because parties recognize the person on the 

other end of the conversation. Even if the conversation is with a stranger, concern 

with source authentication applies primarily to call setup, rather than to the 

authentication of voice packets in the midst of a conversation. As a result of these 

considerations, some designers may consider HMAC less important for secure 

VOIP than call encryption, and may limit HMAC use if performance is a 

problem. 

38 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

5 SIP 

SIP is the IETF specified protocol for initiating a two-way communication 

session. It is considered by some to be simpler than H.323 [18][16], though it is 

now the largest RFC in IETF history. SIP is text based; thereby avoiding the 

ASN.1 associated parsing issues that exist with the H.323 protocol suite, if 

S/MIME is not used as part of SIP inherent security measures.. Also, SIP is an 

application level protocol, that is, it is decoupled from the protocol layer it is 

transported across. It can be carried by TCP, UDP, or SCTP. UDP may be used 

to decrease overhead and increase speed and efficiency, or TCP may be used if 

SSL/TLS is incorporated for security services. Newer implementations may use 

stream control transmission protocol (SCTP), developed in the IETF SIGTRAN 

working group (RFC 2960) specifically to transport signaling protocols. SCTP 

offers increased resistance to DoS attacks through a four-way handshake method, 

the ability to multi-home, and optional bundling of multiple user messages into a 

single SCTP packet. Additional security services can be used with SCTP via RFC 

3436 (TLS over SCTP) or 3554 (SCTP over IP Sec). Unlike H.323, only one port 

is used in SIP (note that H.323 may also be used in a way that uses only one port 

– direct routed calls). The default value for this port is 5060. 

5.1 SIP Architecture 

The architecture of a SIP network is different from the H.323 structure. A SIP 

network is made up of end points, a proxy and/or redirect server, location server, 

and registrar. A diagram is provided in Figure 5. In the SIP model, a user is not 

bound to a specific host (neither is this the case in H.323, gatekeeper provides 

address resolution). The user initially reports their location to a registrar, which 

may be integrated into a proxy or redirect server. This information is in turn 

stored in the external location server. 

Messages from endpoints must be routed through either a proxy or redirect server. 

The proxy server intercepts messages from endpoints or other services, inspects 

their “To:” field, contacts the location server to resolve the username into an 

address and forwards the message along to the appropriate end point or another 

server. Redirect servers perform the same resolution functionality, but the onus is 

placed on the end points to perform the actual transmission. That is, Redirect 

servers obtain the actual address of the destination from the location server and 

return this information to the original sender, which then must send its message 

directly to this resolved address (similar to H.323 direct routed calls with 

gatekeeper).

The SIP protocol itself is modeled on the three-way handshake method 

implemented in TCP (see Figure 6). We will consider the setup here when a 

proxy server is used to mediate between endpoints. The process is similar with a 

redirect server, but with the extra step of returning the resolved address to the 

source endpoint. During the setup process, communication details are negotiated 

39 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

between the endpoints using Session Description Protocol (SDP), which contains 

fields for the codec used, caller’s name, etc. If Bob wishes to place a call to Alice 

he sends an INVITE request to the proxy server containing SDP info for the 

session, which is then forwarded to Alice’s client by Bob’s proxy, possibly via 

her proxy server. Eventually, assuming Alice wants to talk to Bob, she will send 

an “OK” message back containing her call preferences in SDP format. Then Bob 

will respond with an “ACK”. SIP provides for the ACK to contain SDP instead 

of the INVITE, so that an INVITE may be seen without protocol specific 

information. After the “ACK” is received, the conversation may commence along 

the RTP / RTCP ports previously agreed upon. Notice that all the traffic was 

transported through one port in a simple (text) format, without any of the 

complicated channel / port switching associated with H.323. Still, SIP presents 

several challenges for firewalls and NAT. These difficulties are discussed in the 

next section. 

Figure 5. SIP Network Architecture 

 

5.2 Existing SecurityFeatures within theSIP Protocol 

RFC 3261 describes several security features for SIP, which will be described in 

the next subsections. RFC 3261 deprecates several security features, which were 

advocated in the original RFC 2543, such as the usage of PGP and HTTP Basic 

Authentication. 

40 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

5.2.1 Authentication of Signaling Data using HTTP Digest Authentication 

The Digest authentication scheme is based on a simple challenge-response 

paradigm. The digest authentication scheme challenges the remote end using a 

nonce value. SIP digest authentication is based on the digest authentication 

defined in RFC 2617. Here, a valid response contains a checksum (by default, the 

MD5 checksum) of the user name, the password, the given nonce value, the 

HTTP method, and the requested URI. In this way, the password is never sent in 

the clear. Because of its weak security, and to avoid attacks by downgrading the 

required security level of the authentication, HTTP Basic Authentication was not 

recommended in the current draft of RFC 3261. 

 

5.2.2 S/MIME Usage within SIP 

SIP messages carry MIME bodies. MIME itself defines mechanisms for the 

integrity protection and the encryption of the MIME contents. SIP may use 

S/MIME to enable mechanisms like public key distribution, authentication and 

integrity protection, or confidentiality of SIP signaling data. S/MIME may be 

considered as a replacement for PGP to provide means for integrity protection and 

encryption of SIP messages. To be able to protect SIP header fields as well, 

tunneling of SIP messages in MIME bodies is specified. Generally the proposed 

SIP tunneling for SIP header protection will create additional overhead. S/MIME 

requires certificates and private keys to be used, whereas the certificates may be 

issued by a trusted third party or may be self-generated. The latter case may not 

provide real user authentication but may be used to provide a limited form of 

message integrity protection. The following sections explain the usage of 

S/MIME more deeply. 

The current document, RFC 3261, recommends S/MIME to be used for UAs. 

Moreover, if S/MIME is used to tunnel messages (described below) it is 

recommend using a TCP connection because of the larger messages. This is to 

avoid problems that may arise by the fragmentation of UDP packets. The 

following services can be realized: (a) authentication and integrity protection of 

signaling data, and (b) confidentiality of signaling data 

5.2.3 Confidentiality of Media Data 

SIP itself does not consider the encryption of media data. Using the RTP 

encryption as defined in RFC 1889 may provide confidentiality for media data. 

Another option for media stream security is the use of SRTP [DSRTP]. For key 

management SDP (cf. RFC 2327) may be used. SDP can convey session keys for 

media streams. Note that using SDP for the key exchange provides no method to 

send an encrypted media stream key (cf. Appendix A2). Therefore, the signaling 

request should be encrypted, preferably by using End-to-End encryption. 

41 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

5.2.4 TLS usage within SIP 

RFC 3261 mandates the use of TLS for proxies, redirect servers, and registrars to 

protect SIP signaling. Using TLS for UAs is recommended. TLS is able to protect 

SIP signaling messages against loss of integrity, confidentiality and against 

replay. It provides integrated key-management with mutual authentication and 

secure key distribution. TLS is applicable hop-by-hop between UAs/proxies or 

between proxies. The drawback of TLS in SIP scenarios is the requirement of a 

reliable transport stack (TCP-based SIP signaling). TLS cannot be applied to 

UDP-based SIP signaling. Just as secure HTTP is specified with the “https:”, 

secure SIP is specified with a Universal Resource Indicator (URI) that begins with 

“sips:”. 

 

5.2.5 IPsec usage within SIP 

IPsec may also be used to provide security for SIP signaling at the network layer. 

This type of security is most suited to securing SIP hosts in a SIP VPN scenario 

(SIP user agents/proxies) or between administrative SIP domains. IPsec works 

for all UDP, TCP and SCTP based SIP signaling. IPsec may be used to provide 

authentication, integrity and confidentiality for the transmitted data and supports 

end-to-end as well as hop-by-hop scenarios. At this time there is no default cipher 

suite for IPsec defined in SIP. Note that RFC 3261 does not describe a 

framework for the use of IPsec and no requirement is given as to how the key 

management is to be realized, or which IPsec header and mode is to be used. One 

accepted protocol for key management is Internet Key Exchange (IKE), a hybrid 

protocol based on the Internet Security Association and Key Management 

Protocol (ISAKMP), the Oakley Key Determination Protocol (RFC 2412) and the 

Secure Key Exchange Mechanism for the Internet (SKEME). The IKE protocol 

provides automated cryptographic key exchange and management mechanisms

for IPsec. IKE is used to negotiate security associations (SAs) for use with its own 

key management exchanges (called Phase 1) and for other services such as IPsec 

(called Phase 2). IKE is particularly used in the establishment of VPNs. 

5.2.6 Security Enhancements for SIP 

Currently within the IETF several drafts concerning security are being discussed, 

with a view toward providing a general security solution to SIP scenarios. 

Several drafts have been produced concerning authentication, integrity, and 

confidentiality for SIP. The following subsections provide a short overview of 

Internet drafts, which may be of interest for a discussion of security enhancements 

for common SIP scenarios. This list of Internet drafts is not complete, as this is a 

continually evolving area, but the most significant drafts are considered here. 

42 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

5.2.6.1 SIP Authenticated Identity Body 

SIP Authenticated Identity Body (AIB) defines a generic SIP authentication 

token. The token is provided by adding an S/MIME body to a SIP request or 

response in order to provide reference integrity over its headers. The document 

defines a format for this message body referred to as an authenticated identity 

body (AIB). This is a digitally signed SIP message (sip/message) or message 

fragment (sip/frag). 

 

5.2.6.2 SIP Authenticated Identity Management 

The existing mechanisms for expressing identity in SIP often do not permit an 

administrative domain to securely verify the identity of the originator of a request. 

This document recommends practices and conventions for authenticating end 

users, and proposes a way to distribute cryptographically secure authenticated 

identities within SIP messages by including an authentication token (as a MIME 

body). This token is then added to the message. 

5.2.6.3 S/MIME AES Requirement for SIP 

RFC 3261 specifies 3DES as the required minimum encryption algorithm for 

implementations of S/MIME in SIP. Although 3DES is still a viable algorithm,

NIST has selected an improved algorithm, AES, as a replacement for DES and 

3DES. Standards-track RFC 3853, the S/MIME AES Requirement for SIP, 

updates the normative guidance of RFC 3261 to require AES for S/MIME. This 

specification may be found at: http://www.ietf.org/rfc/rfc3853.txt. AES provides 

higher throughput and lower computational complexity than 3DES, and can be 

implemented with low memory requirements, making it more suitable for mobile 

or embedded devices, including VOIP phones.

5.2.6.4 Security Mechanism Agreement for SIP 

SIP has a number of security mechanisms. Some of them have been built in to the 

SIP protocol directly, such as HTTP authentication. These mechanisms have 

alternative algorithms and parameters. The idea originates from the 3rd Generation 

Partnership Project (3GPP), a collaboration of telecommunications companies, 

and provides a mechanism for selecting which security mechanisms to use 

between two entities. RFC 3261 itself does not provide any mechanism agreement 

options. Moreover, even if some mechanisms such as OPTIONS were used to 

perform a mechanism agreement, the agreement would be vulnerable to BiddingDown attacks (a phase of man-in-the-middle attack where the attacker modifies 

messages to convince communicating parties that both sides support only weak 

algorithms). Three header fields are defined for negotiating the security 

mechanisms within SIP between a SIP User Agent entity and its next hop SIP 

server. It is a proposed standard (RFC 3329) from the IETF. Five mechanisms

are currently supported: 

43 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

− TLS 

− HTTP Digest 

− IPsec with IKE 

− manually keyed IPsec without IKE 

− S/MIME 

5.2.6.5 End-to-Middle, Middle-to-Middle, Middle-to-End Security 

Currently there are two drafts being discussed within the IETF dealing with endto-middle, middle-to-middle and middle-to- end security: “End-to-middle 

Security in the Session Initiation Protocol (SIP)” [30] and “A Mechanism to 

Secure SIP information inserted by Intermediaries” [31]. The motivation for the 

End to Middle Security draft [30] stems from the need to enable intermediaries to 

use some of the SIP message header and body when end-to-end security is 

applied. Examples include logging services for enterprise use, firewall traversal, 

and transcoding (tailoring web pages for varying devices such as PDAs or cell 

phones). Intermediaries may not be able to trace the SIP message body for certain 

information (e.g., port numbers to be opened) if the body is encrypted. This draft 

is related to [31]. There is still a discussion about this draft within the SIPPING 

group. 

The second draft [31] aims at a mechanism to secure information inserted by 

intermediaries. This document came about to provide a more robust security 

solution for History-Info header, however, the intention is to provide an overall 

more robust security solution for SIP. Proxies sometimes have the need to read 

and/or modify a message body or header in a request. However, this is explicitly 

precluded by the SIP specification and is further complicated when a message 

body is protected with S/MIME. RFC 3261 is designed so that a proxy does not 

break integrity of the body. 

The security requirements between both approaches are slightly different, since 

information is added by intermediaries and used by intermediaries. Nevertheless, 

SIP End to Middle Security [30] and SIP Intermediate Security [31] share the 

same fundamental problems to be solved in SIP. It is anticipated that there will be 

further discussion on this item, as certain scenarios exist where this functionality 

is needed. 

5.2.7 SIP Security Issues 

The text encoding of SIP makes it easier to analyze using standard parsing tools 

such as Perl or lex and yacc. Still, some new requirements are placed on the 

firewall in a SIP-based VOIP network. First, firewalls must be stateful and 

monitor SIP traffic to determine which RTP ports are to be opened and made 

available to which addresses. This responsibility is similar to the task firewalls on 

44 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

an H.323 based network perform, except the call setup and header parsing is 

much simpler. The other issues SIP-based VOIP encounters with firewalls are 

associated with RTP traffic and incoming calls, as covered in section 7.4. As with 

H.323, the big problem for SIP is NAT. 

NAT inhibits SIP’s registration and communication mechanisms and requires 

innovative solutions to resolve these issues. The problems exist because in a SIPbased network, the SIP proxy is normally outside the NAT device. There are 

three main scenarios for using a SIP proxy: 

− The proxy is within the corporate LAN and the Teleworker connects from 

outside 

− The proxy is at the telecom side and clients from, for instance, smaller 

companies connecting to this proxy for VOIP service 

− Two administrative domains are connected, both have their own proxy. 

So the problem is bartering communication between a proxy server that deals with 

global IP addresses and a machine that has been assigned a private network 

address. Rosenberg & Schulzrinne [22] classify three different sets of problems 

SIP traffic has in such an architecture: originating requests, receiving requests, 

and handling RTP. We have already dealt with the incompatibilities of RTP with 

NAT and now we will see the issues NAT presents to the call setup process itself.

To initialize a session from behind the NAT, a caller can simply send an INVITE 

message as always. The outgoing port number (5060) will be preserved by the 

NAT, but response communication could be disturbed. If SIP is implemented 

over UDP (recall SIP is protocol independent) the proxy server must send the 

UDP response to the address and port the request arrived on [22] A simpler 

solution is to use the standard practice of routing SIP communication over TCP. 

With TCP, the response from the callee will come over the same channel as the 

original INVITE and so NAT will not present a problem. 

45 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

Bob Calls Alice in an SIP Network (Protocol View)

CONVERSATION

Bob Proxy Server(s) Alice

INVITE Alice

INVITE Alice@123.45.67.8

OK from Alice@123.45.67.8

ACK Alice with route

Alice@123.45.67.8

ACK Alice@123.45.67.8

OK from Alice@123.45.67.8

BYE Alice@123.45.67.8

BYE Alice@123.45.67.8

OK

OK

Figure 6. SIP Protocol

We have already discussed some of the problems with incoming VOIP 

connections against NAT (see Section 4.2). Now we will look more in depth at 

the SIP specific problems with incoming calls. Rosenberg & Schulzrinne [22] 

trace the problem back to the registration process itself. When a user contacts the 

registrar, they provide their IP address as their reachable address and this is stored 

in the location server. However, this is their private IP address. The proxy server 

deals only with global IP addresses, so when a message comes in for 

username@domain.com, it will attempt to route this call to the registered address, 

but in the public domain. For instance, if username@domain.com is registered to 

an internal IP address of 10.7.34.189, then the proxy server will attempt to 

forward the traffic to this address, but in the public domain. This address is 

unreachable for the proxy server and the connection will be refused. The solution 

to this is a delicate manipulation of IP addresses and an expansion of the 

responsibilities of the SIP proxy server. 

46 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

6 Gateway Decomposition 

Media gateway control protocols address the requirements of IP telephony 

networks that are built using “decomposed” VOIP gateways. Decomposed VOIP 

gateways consist of Media Gateways (MGs) and Media Gateway Controllers 

(MGC), and appear to the outside as a single VOIP gateway. MGC handles the 

signaling data between the MGs and other network components such as H.323 

gatekeepers or SIP Servers, or towards SS7 Signaling Gateways. MGs focus on 

the audio signal translation function, performing conversion between the audio 

signals carried on telephone circuits and data packets carried over the Internet or 

other packet networks. A single MGC can control multiple MGs, which leads to 

cost reductions when deploying larger systems. Common examples are the Media 

Gateway Control Protocol (MGCP) and Megaco/H.248, which are described 

within the next subsections. 

6.1 MGCP 

6.1.1 Overview 

MGCP is used to communicate between the separate components of a 

decomposed VOIP gateway. It is a complementary protocol to SIP and H.323. 

The MGCP protocol was derived from version 1.1 of the SGCP protocol, which 

was a fusion of the SGCP version 1 and IPDC. MGCP is currently being 

maintained by PacketCable (called NCS (Network Call Signaling Protocol)) and 

the Softswitch Consortium. In October 1999, MGCP was converted into an 

informational RFC 2705. 

There are plans for the MGCP specification to be enhanced by international 

standards bodies. One is the IP Cablecom activity proposing J.162 (Network Call 

Signaling) and J.171 (Trunking Gateway Control Protocol, a variant of J.162). A 

similar version of these proposals will also be provided within ETSI as 

EuroPacketCable specifications. At the present time, MGCP is the de-facto 

industry standard and has not yet been superseded by MEGACO/H.248. 

47 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

6.1.2 System Architecture 

Within MGCP the MGC server or “call agent” is mandatory and manages calls 

and conferences, and supports the services provided (see Figure 7). The MG 

endpoint is unaware of the calls and conferences and does not maintain call states. 

MGs are expected to execute commands sent by the MGC call agents. MGCP 

assumes that call agents will synchronize with each other sending coherent 

commands to MGs under their control. MGCP does not define a mechanism for 

synchronizing call agents. MGCP is a master/slave protocol with a tight coupling 

between the MG (endpoint) and MGC (server). 

Client

Media Gateway Controller

MGC (Call Agent)

Media

Gateway

MGCP

H.323 / SIP

IP IP

Media

Gateway

PSTN

LAN

RTP

Figure 7: General Scenario for MGCP Usage 

RTP data is exchanged directly between the involved media gateways. The Call 

Agent uses MGCP to provide the gateways with the description of connection 

parameters such as IP addresses, UDP port and RTP profiles. These descriptions 

follow the conventions delineated in the Session Description Protocol (SDP) from

RFC 2327. SDP is a session description protocol for multimedia sessions, which 

runs also on UDP connections. 

6.1.3 Security Considerations 

There are no security mechanisms designed into the MGCP protocol itself. The 

informational RFC 2705 refers to the use of IPsec (either AH or ESP) to protect 

MGCP messages. Without this protection a potential attacker could set up 

unauthorized calls or interfere with ongoing authorized calls. Beside the use of 

IPsec, MGCP allows the call agent to provide gateways with session keys that can 

be used to encrypt the audio messages, protecting against eavesdropping. The 

session key will be used later on in RTP encryption. The RTP encryption,

described in RFC 1889, may be applied. Session keys may be transferred between 

the call agent and the gateway by using the SDP (cf. RFC 2327). 

48 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

6.2 Megaco/H.248

6.2.1 Overview 

The IETF started to work on MEGACO as a compromise protocol between 

MGCP and MDCP. ITU-T SG16 adopted MEGACO version 0.1 in April 1999 as 

the starting specification for H.GCP (H-series, Gateway Control Protocol), later 

H.248. In June 1999, the IETF MEGACO WG and ITU-T came out with a single 

document describing a standard protocol for interfacing between Media Gateway 

Controllers (MGCs) and Media Gateways (MGs) MEGACO/H.248. 

MEGACO/H.248. It is expected to win wide industry acceptance as the official 

standard for decomposed gateway architectures released by both the IETF and 

ITU-T. 

Since MEGACO/H.248 is derived from MGCP, many similarities can be found, 

for instance: 

– Similarity between the semantics of the commands in the two specifications. 

– The use of ABNF grammar for syntax specification and the Session 

Description Protocol (SDP) to specify media stream properties is the same as 

in MGCP. 

– The processing of signals and events in media streams is the same in 

MEGACO as in MGCP. 

– The concept of packages containing event and signal definitions that permits 

easy extension to the protocol is borrowed from MGCP. 

– The MEGACO specification for transport of messages over UDP is the same

as specified in MGCP. The three-way-handshake and the computation of 

retransmission timers described in MGCP are also described within the ALF 

definition specified in Annex E of MEGACO. 

MEGACO/H.248 introduces several enhancements compared with MGCP, 

including the following: 

– Support of multimedia and multipoint conferencing enhanced services 

– Improved syntax for more efficient semantic message processing 

– TCP and UDP transport options 

– Allows either text or binary encoding (to support IETF and ITU-T approach) 

– Formalized extension process for enhanced functionality 

– Expanded definition of packages 

MEGACO is described as Gateway Control Protocol Version 1 within the RFC 

3525. 

6.2.2 System Architecture 

MEGACO/H.248 (see Figure 8) has basically the same architecture as MGCP. 

MEGACO/H.248 commands are similar to MGCP commands. However, the 

49 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

protocol models are quite different. MEGACO specifies a media gateway 

connection model that has two entities: Terminations (source or sink for (one or 

more) media streams), and context (grouping of terminations connected in a call). 

In contrast, MGCP uses the following two entities: Endpoints (source or sink of 

data), and Connection: (association between two endpoints). 

Taking a multipoint conference as an example, MEGACO simplifies the 

connection setup by adding terminations to a context, whereas MGCP has to 

establish several connections to the conference server. The context in this scenario 

may cover multiple media streams for enhanced multimedia services. 

With MEGACO/H.248, the primary mechanism for extension is by means of 

packages. In general, MEGACO/H.248 Packages include more detail than MGCP 

Packages. They define additional properties and statistics along with event and 

signal information that may occur on terminations. 

Client

Media Gateway Controller

MGC (Call Agent)

Media

Gateway

MEGACO / H.248

H.323 / SIP

IP IP

Media

Gateway

PSTN

LAN

RTP

Figure 8: General Scenario for MEGACO/H.248 Usage

6.2.3 Security Considerations 

Megaco (RFC 3525) recommends security mechanisms that may be in underlying 

transport mechanisms, such as IPsec. H.248 goes a step further by requiring that 

implementations of the H.248 protocol implement IPsec if the underlying 

operating system and the transport network support IPsec. Implementations of the 

protocol using IPv4 are required to implement the interim AH scheme. H.248 

states that implementations employing the AH header shall provide a minimum 

set of algorithms for integrity checking using manual keys (compliant to RFC 

2402). 

The interim AH scheme is the use of an optional AH header, which is defined in 

the H.248 protocol header. The header fields are exactly those of the SPI, 

50 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

SEQUENCE NUMBER and DATA fields as defined in RFC 2402. The semantics 

of the header fields are the same as the "transport mode" of RFC 2402, except for 

the calculation of the Integrity Check Value. For more details on the calculation 

of the ICV check H.248. The interim AH interim scheme does not provide 

protection against the eavesdropping and replay attacks. 

For MEGACO, manual key management is assumed and replay protection, 

defined for IPsec, may not be used in this scenario (the sequence number in the 

AH may overrun when using manual key management, since re-keying is not 

possible). Furthermore, H.248 states that implementations employing the ESP 

header shall provide a minimum set of algorithms for integrity checking and 

encryption (compliant to RFC 2402). Moreover, implementations should use IKE 

(RFC 2409) to permit more robust keying options. Implementations employing 

IKE should support authentication with RSA signatures and RSA public key 

encryption. 

51 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

7 Firewalls, Address Translation, and Call Establishment 

Firewalls and NAT present a formidable challenge to VOIP implementers. 

However, there are solutions to these problems, if one is willing to pay the price. 

Commonly used solutions that can be integrated into a standard network 

configuration containing a firewall and/or NAT are presented here. Much of the 

standards work thus far has been done for SIP in the IETF, though the ITU has 

recently taken up the firewall/NAT traversal issue for H.323 in Study Group 16. 

It is important to note that all three major VOIP protocols, SIP, H.323, and 

H.248/MEGACO all have similar problems with firewalls and NATs. Although 

the use of NATs may be reduced as IPv6 is adopted, they will remain a common 

component in networks for years to come, and IPv6 will not alleviate the need for 

firewalls, so VOIP systems must deal with the complexities of firewalls and 

NATs. 

7.1 Firewalls 

Firewalls are a staple of security in today’s IP networks. Whether protecting a 

LAN, WAN, encapsulating a DMZ, or just protecting a single computer, a 

firewall is usually the first line of defense against would be attackers. Firewalls 

work by blocking traffic deemed to be invasive, intrusive, or just plain malicious 

from flowing through them. If networks are castles, firewalls are the 

drawbridges. Traffic not meeting the requirements of the firewall is dropped. 

Processing of traffic is determined by a set of rules programmed into the firewall 

by the network administrator. These may include such commands as “Block all 

FTP traffic (port 21)” or “Allow all HTTP traffic (port 80)”. Much more complex 

rule sets are available in almost all firewalls. 

A useful property of a firewall, in this context, is that it provides a central location 

for deploying security policies. It is the ultimate bottleneck for network traffic 

because when properly designed, no traffic can enter or exit the LAN without 

passing through the firewall. This situation lends itself to the VOIP network 

where firewalls simplify security management by consolidating security measures 

at the firewall gateway, instead of requiring all the endpoints to maintain up to 

date security policies. This takes an enormous burden off the VOIP network 

infrastructure. Note that this abstraction and simplification of security measures 

comes at a price. The introduction of firewalls to the VOIP network complicates 

several aspects of VOIP, most notably dynamic port trafficking and call setup 

procedures. This chapter describes various complications firewalls introduce into 

the system. But since firewalls are an essential and often already deployed 

component of the modern network, we will also examine some proposed and 

applied solutions to these entanglements. 

52 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

7.1.1 Stateful Firewalls

Most VOIP traffic travels across UDP ports. Firewalls typically process such 

traffic using a technique called packet filtering. Packet filtering investigates the 

headers of each packet attempting to cross the firewall and uses the IP addresses, 

port numbers, and protocol type (collectively known as the 5-tuple) contained 

therein to determine the packets’ legitimacy. In VOIP and other media streaming 

protocols, this information can also be used to distinguish between the start of a 

connection and an established connection. There are two types of packet filtering 

firewalls, stateless and stateful. Stateless firewalls retain no memory of traffic 

that has occurred earlier in the session. Stateful firewalls do remember previous 

traffic and can also investigate the application data in a packet. Thus, stateful 

firewalls can handle application traffic that may not be destined for a static port. 

7.1.2 VOIP specific Firewall Needs 

In addition to the standard firewall practices, firewalls are often deployed in VOIP 

networks with the added responsibility of brokering the data flow between the 

voice and data segments of the network. This is a crucial functionality for a 

network containing PC-Based IP phones that are on the data network, but need to 

send voice messages. All voice traffic emanating from or traveling to such 

devices would have to be explicitly allowed in if no firewall was present because 

RTP makes use of dynamic UDP ports (of which there are thousands). Leaving 

this many UDP ports open is an egregious breach of security. Thus, it is 

recommended that all PC-based phones be placed behind a stateful firewall to 

broker VOIP media traffic. Without such a mechanism, a UDP DoS attack could 

compromise the network by exploiting the plethora of open ports. This is one 

example of how firewalls are used to provide a logical segmentation of the 

network, providing a barrier between voice and data sectors. Halpern [23] 

identifies some of the key points of collision between voice and data traffic where 

firewalls are necessary, including: 

• 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. 

Halpern [23] provides details for all these connections. Firewalls could also be 

used to broker traffic between physically segmented traffic (one network for 

VOIP, one network for data) but such an implementation is fiscally and physically 

unacceptable for most organizations, since one of the benefits of VOIP is voice 

and data sharing the same physical network. 

53 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

7.2 NetworkAddress Translation

Network Address Translation (NAT) is a powerful tool that can be used to hide 

internal network addresses and enable several endpoints within a LAN to share 

the same (external) IP address. For the purposes of this document, NAT actually 

refers to Network Address and Port Translation (NAPT). In NAT as it is literally 

defined, outgoing IP headers are changed from private LAN addresses to the 

router’s global IP. In NAPT, the TCP/UDP headers themselves are converted. 

This allows several computers to simultaneously share the router’s global IP 

address. Also, machines that do not need to access the Internet can still be 

assigned local addresses on the intranet without producing conflicts or needlessly 

taking up an IP address. With the shortages of IP addresses in many regions, this 

is an extremely useful functionality. 

NATs also indirectly contribute to security for a LAN, making internal IP 

addresses less accessible from the public Internet. Thus, all attacks against the 

network must be focused at the NAT router itself. Like firewalls, this provides 

security because only one point of access must be protected, and the router will 

generally be far more secure than a PC directly connected to the Internet (less 

likelihood of open ports, malicious programs, etc.). The abstraction of the LAN 

from the Internet through a NAT also simplifies network management. For 

instance, if one decided to change their ISP, only the external router configuration 

would need to be changed. The internal network and addressing scheme could be 

left untouched [24]. 

Different types of NAT policies result in different complexity (the following 

terminology is adopted from the STUN RFC [32] describing a NAT traversal 

methodology, which originates from the MIDCOM working group of the IETF): 

Full Cone NAT 

A full cone NAT is one where all requests from the same internal IP 

address and port are mapped to the same external IP address and port. 

Furthermore, any external host can send a packet to the internal host, by 

sending a packet to the mapped external address. 

Restricted Cone NAT

Restricted Cone: A restricted cone NAT is one where all requests from the 

same internal IP address and port are mapped to the same external IP 

address and port. Unlike a full cone NAT, an external host (with IP 

address X) can send a packet to the internal host only if the internal host 

had previously sent a packet to IP address X. 

54 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

Port Restricted Cone 

A port restricted cone NAT is like a restricted cone NAT, but the 

restriction includes port numbers. Specifically, an external host can send a 

packet, with source IP address X and source port P, to the internal host 

only if the internal host had previously sent a packet to IP address X and 

port P. It is used to enable sharing of external IP addresses. 

Symmetric NAT 

A symmetric NAT is one where all requests from the same internal IP 

address and port, to a specific destination IP address and port, are mapped 

to the same external IP address and port. If the same host sends a packet 

with the same source address and port, but to a different destination, a 

different mapping is used. Furthermore, only the external host that 

receives a packet can send a UDP packet back to the internal host. 

All of these benefits of NAT come at a price. NATs “violate the fundamental 

semantic of the IP address, that it is a globally reachable point for 

communications” [25]. This design has significant implications for VOIP. For 

one thing, an attempt to make a call into the network becomes very complex when 

a NAT is introduced. The situation is analogous to a phone network where 

several phones have the same phone number, such as in a house with multiple 

phones on one line (see 

Figure 9). There are also several issues associated with the transmission of the 

media itself across the NAT, including an incompatibility with IPsec detailed in 

section 8.4. 

Figure 9. IP Telephones Behind NAT and Firewall 

Conceptually, the easiest solution to these incompatibilities is to do away with 

55 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

NATs entirely, but NATs have their benefits, and even if IPv6 and its expanded 

address space were implemented today and enough IP addresses were available 

for everyone to have their own unique IP’s, there would still be a need for NATs. 

Some ISPs use a scheme where users are assigned static IP addresses, one per 

user. It is unlikely that an ISP would completely overhaul its system and move to 

a dynamic IP assignment (i.e. DHCP) just because a wealth of new addresses are 

available to IPv6. This would undermine their whole network and lead to 

vulnerabilities and opportunities for malicious users to steal Internet access. But 

many users will still want to connect multiple machines to the Internet using a 

single IP address, and so the use of NATs will continue. There are many 

scenarios analogous to this one where NATs are both the cheapest, easiest, and 

most efficient solution, so NATs are not likely to be abandoned. 

 

7.3 Firewalls,NATs, and VOIPIssues

Some VOIP issues with firewalls and NATs are unrelated to the call setup

protocol used. Both network devices make it difficult for incoming calls to be 

received by a terminal behind the firewall / NAT. Also, both devices affect QoS 

and can wreak havoc with the RTP stream. The following sections describe these 

non-protocol specific issues. 

7.3.1 Incoming Calls 

Regardless of the protocol used for call setup, firewalls and NATs present 

considerable difficulties for incoming calls. Allowing signal traffic through a 

firewall from an incoming call means leaving several ports open that might be 

exploited by attackers. Careful administration and rule definitions should be used 

if holes are to be punched in the firewall allowing incoming connections. 

Solutions exist without such holes, including Application Level Gateways and 

Firewall Control Proxies. NAT creates even more difficulties for incoming calls. 

Any IP application, including VOIP, that needs to make a connection from an 

external realm to a point behind a NAT router, would need to know this point’s 

external IP and port number assigned by the router [10]. This situation is far from

ideal because it precludes a caller outside the NAT from reaching an internal 

address except in extreme circumstances. In fact, with dynamic ports being 

assigned by the NAT, this is nearly an impossible situation because the port the 

caller requests will be changed by the NAT. Thus, an IP telephony endpoint 

behind a NAT being analogous to a phone behind a switchboard such that it can 

only make outgoing calls. For endpoints behind firewalls and NATs, it may be 

necessary to publish the contact address to enable other clients to call them. 

This is not an acceptable solution for thousands of people using NAT today. 

However, there are some solutions to this problem (see chapter 8). 

56 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

7.3.2 Effects on QoS 

Both firewalls and NATs can degrade QoS in a VOIP system by introducing 

latency and jitter. NATs can also act as a bottleneck on the network because all 

traffic is routed through a single node. QoS related issues stemming from these 

network staples are presented here. 

VOIP is highly sensitive to latency. So a firewall needs to be able to broker data 

traffic, but cannot incur time penalties of any significant length, even those that 

would go unnoticed with simple data traffic [26]. At issue is not only how fast 

the firewall can interact with the network traffic, but how fast its processor can 

handle VOIP packets. Two aspects of VOIP can cause degraded behavior in the 

firewall CPU. First, the call setup process has to be done using H.323 or SIP (see 

chapters 4-5). Regardless of the protocol used, firewalls need to “dig deeper” into 

these packets to determine their validity. A flood of call request packets, as the 

result of an increase in call volume or a malicious attack, can intensify this effect. 

The presence of a NAT compounds this issue because the payload of the packet 

must then be changed at the application level to correspond to the NAT translated 

source or destination address and ports, requiring not only “digging” but filling in 

the hole with new dirt as well. All this labor puts a tremendous burden on the 

firewall processor, which must accomplish all these tasks while introducing the 

bare minimum in latency, especially if protocol security measures are used, such 

as message integrity. 

The other aspect of VOIP that can put a strain on a firewall CPU is the small but 

plentiful number of RTP packets that make up a VOIP conversation. Firewalls 

are rarely concerned with the size of a packet, but since each packet must be 

inspected, a large number of packets can stress the firewall. For example, a 

firewall may support 100 Mbit/sec (based on the assumption of large packets), but 

may be overloaded by a flood of small 50 byte packets long before the 100 

Mbit/sec rate is reached [LaCour qtd in 26]. QoS/VOIP aware firewalls are 

designed to avoid performance problems such as these. 

No matter how fast the network connection, the firewall CPU is a bottleneck for 

all unencrypted network packets. Thus a solution to this issue is to use a VPN for 

all VOIP traffic. Discussion of the pros and cons of this choice is presented in 

chapter 8. 

7.3.3 Firewalls and NATs 

Firewalls have difficulties sorting through VOIP signaling traffic. There are 

solutions to this but there is an additional, and even more vexing problem

associated with firewalls and VOIP media. RTP traffic is dynamically assigned 

an even port number in the range of UDP ports (1024-65534). In addition, the 

RTCP port controlling this stream will flow through an associated, randomlyassigned port. Allowing such traffic along such a vast number of ports by default 

would leave the system highly exposed. So firewalls must be made aware 

57 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

dynamically of which ports media is flowing across and between which terminals. 

For this reason, only stateful firewalls that can process H.323 and SIP should be 

incorporated into the network to open and close ports. Many new firewalls come

equipped with such functionality, although sometimes they support only one 

protocol (H.323 or SIP) [see 27 for a partial list]. If such firewalls are not 

available/feasible there are additional hardware solutions available, or VPNs can 

be used to tunnel through the firewall (chapter 8) 

NATs also introduce major design complications into media traffic control in 

VOIP. First of all, the standard NAT practice of assigning new port numbers at 

random breaks the pair relationship of RTP and RTCP port numbers [7]. The 

translation of IP Addresses and ports by NAT is also problematic for the reception 

of VOIP packets. If the NAT router does not properly process the traffic, the new 

addresses/ports will not correspond to those negotiated in the call setup process. 

In this scenario, the VOIP gateway may not properly deliver the RTP packets. 

The problem is exacerbated if both call participants are behind NATs. Several 

solutions are available for users with administrative control of their network 

security and the money to upgrade their firewall/NAT hardware (see 7.1-7.2). For 

users without control of their network architecture, there is a proposed solution 

[25], which is detailed in section 9.6. However, this solution breaks down when 

both users are behind NATs and it has not been implemented successfully to date. 

The use of NATs adds another possible complication to VOIP call signaling due 

to the finite nature of NAT bindings. At a NAT, a public IP address is bound to a 

private one for a certain period of time (t). This entry gets deleted if no traffic was 

observed at the NAT for t seconds or the connection was torn down explicitly. A 

SIP INVITE message, which triggers the binding of the private address to the 

public one, establishes “state” information when it passes through NATs or 

firewalls. This state eventually must be cleared. When TCP is used, closure of the 

TCP connection is usually a good indicator of the termination of the 

application. However, when SIP runs over UDP such a clear indication is 

missing. Furthermore, as a silence period during a conversation might be longer 

than t seconds, not receiving traffic for t seconds might not suffice as an 

indication of session termination. As a result, it is possible that some state 

information is destroyed before the transaction and/or the call actually completes. 

7.4 CallSetup Considerationswith NATs and Firewalls 

VOIP users will not tolerate excessive latency in the call setup process, which 

corresponds to lifting the receiver and dialing in a traditional system. Users may 

be annoyed with a setup process that requires more than a few seconds [12]. 

Many factors influence the setup time of a VOIP call. At the network level, these 

include the topology of the network and the location of both endpoints as well as 

the presence of a firewall or NAT. At the application level, the degree or lack of 

authentication and other data security measures, as well as the choice of protocol 

58 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

used to set up the call, can dramatically alter the time necessary to prepare a VOIP 

connection. 

7.4.1 Application Level Gateways

Application Level Gateways (ALGs) are the typical commercial solution to the 

firewall/NAT traversal problem [10]. An ALG is embedded software on a 

firewall or NAT, that allows for dynamic configuration based on application 

specific information. A firewall with a VOIP ALG can parse and understand 

H.323 or SIP, and dynamically open and close the necessary ports. When NAT is 

employed, the ALG needs to open up the VOIP packets and reconfigure the 

header information therein to correspond to the correct internal IP addresses on 

the private network, or on the public network for outgoing traffic. This includes 

modifying the headers and message bodies (e.g., SDP) in H.323 and SIP. ALG 

implementations are discussed for H.323 in [21] and SIP in [22]. The NAT 

problem is alleviated when the ALG replaces the private network addresses with 

the address of the ALG itself. It works by not only changing the IP address, but 

also mapping RTP traffic into ports the ALG can read from and forward to the 

correct internal machine. The need for consecutive ports for RTP and RTCP can 

cause a problem here [22] because all VOIP traffic on the network (and data 

traffic as well) is being routed through the ALG, so as call volume increases, 

finding enough consecutive ports may become an issue. So although both 

endpoints may have adequate ports to convene a conversation, the firewall’s 

deficiencies may cause the call to be rejected as “busy” by the ALG itself. 

There are significant performance and fiscal costs associated with the 

implementation of an ALG. Performance-wise, the manipulation of VOIP 

packets introduces latency into the system and can contribute to jitter when high 

call volumes are experienced. Depending on the firewall architecture, this can 

also slow down throughput in the firewall, contributing to general network 

congestion. A firewall with ALG support can be expensive, and would need to be 

upgraded or replaced each time the standards for VOIP change. Also, the 

addition of application intelligence to a firewall can introduce instabilities into the 

firewall itself. Some firewalls have been found vulnerable to an attack in which a 

high rate of call setups can be sent, depleting the connection tables of the firewall. 

These half-open VOIP sessions may not time out in the firewall for more than 24 

hours. Still with all these detractions, an ALG remains the simplest and safest 

workaround to allow the coexistence of VOIP, firewalls, and NAT. 

7.4.2 Middlebox Solutions

One drawback to ALGs is that they are embedded in the firewall itself, and thus 

the latency and throughput slowdown of all traffic traversing the firewall is 

aggregated and then compounded by the VOIP call volume. Middlebox-style 

solutions attempt to alleviate this malady by placing an extra device outside the 

firewall that performs many of the functions associated with an ALG. The device 

59 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

that the application intelligence is extracted to can be an “in-path” system such as 

an H.323 gatekeeper or a SIP Proxy that sits in the control path of the session and 

is considered to be a “trusted system” [28] that parses VOIP traffic and instructs 

the firewall to open or close ports based on the needs of the VOIP signaling via a 

midcom protocol (see Figure 10). The midcom protocol has not been finalized 

yet by the IETF. Abstracting stateful inspection and manipulation of signaling 

packets from the NATs and firewalls (middleboxes) will improve scalability and 

in the long run, reduce the cost of updating the network[10] by not having to 

replace the firewall every time the protocols change. There is also a performance 

improvement that comes from abstracting two highly processor intensive tasks 

(VOIP parsing and packet filtering) into two separate spheres of influence. This 

strategy is currently being pursued by the IETF in the Middlebox 

Communications (Midcom) Working Group. 

Figure 10. Middlebox Communications Scenario 

There are some drawbacks to this approach. First, the firewall must be configured 

for control by the application-aware device, which incurs an initial setup cost. 

Also, the middlebox itself requires protection from attackers. A compromised 

midcom agent is disastrous for the network at large because the firewall takes 

control cues from the “trusted” device running the midcom agent. Thus an 

intruder taking control of the midcom agent could open any ports in the firewall 

and then gain access to the private network. So if the application aware device 

(like a SIP Proxy) is placed outside the firewall, a second firewall would have to 

be used to protect that device. 

7.4.3 Session Border Controllers 

While application level gateways may carry scalability concerns, middlebox 

solutions have not found their way out of standards bodies and into commercial 

products as fast as might have been hoped. In the absence of a universally 

accepted solution to the issues associated with firewall/NAT traversal, product 

developers have brought to market a solution that has come to be known as a 

60 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

Session Controller, or a Session Border Controller (SBC). SBCs are dedicated 

appliances that offer one or more of the following services to a VOIP perimeter: 

Firewall/NAT traversal, Call Admission Control, Service Level Agreement 

monitoring, support for lawful intercept, and protocol interworking. Third party 

analysis of these solutions is not widely available as of yet, but in the near term, 

the demand for these products is expected to grow. 

7.5 Mechanisms to solve the NAT problem 

Especially for real-time communication protocols like H.323 and SIP, NAT 

causes trouble because these protocols include IP addresses in their messages. For 

integrity protection the NAT device would have to be a trusted intermediate host 

to recalculate the integrity checksum. From an end-to-end security point of view 

this is not recommended. Therefore the following section describes mechanisms

to handle the NAT problem differently. 

Simple Traversal of UDP through NATs (STUN) 

As defined in RFC 3489, Simple Traversal of User Datagram Protocol (UDP) 

Through Network Address Translators (NATs) (STUN) is a lightweight protocol 

that allows applications to discover the presence and types of NATs and firewalls 

between them and the public Internet. It also provides the ability for applications

to determine the public Internet Protocol (IP) addresses allocated to them by the 

NAT (address bindings). STUN works with many existing NATs, and requires no 

changes to NATs. STUN does not work with symmetric NAT, because the IP 

address port mapping is dependent on the destination, i.e. The STUN server 

would deliver the IP address port mapping for the connection to the STUN server 

itself, which is different from the mapping for the destination client. 

Traversal Using Relay NAT (TURN) 

The draft for Traversal Using Relay NAT (TURN) was submitted by J.Rosenberg 

(dynamicsoft), R. Mahy (Cisco), C. Huitema (Microsoft) in October 2003 and 

revised July 19, 2004. TURN is a protocol that allows for an element behind a 

NAT or firewall to receive incoming data over TCP or UDP connections to 

complement the limitations of STUN. The connection has to be requested by the 

TURN client, TURN supports the connection of a user behind a NAT to only a 

single peer. The TURN server would act as a data relay, receiving data on the 

address it provides to clients, and forwarding them to the clients. TURN is 

identical in syntax and general operation to STUN, but allocates transport address 

bindings. 

Unlike a STUN server, a TURN server provides resources (bandwidth and ports) 

to clients that connect to it. Therefore, only authorized clients can access the 

TURN server. TURN assumes the existence of a long-lived shared secret between 

the client and the TURN server in order to achieve authentication of the TURN 

requests. The client uses this shared secret to authenticate itself in a Shared Secret 

61 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

Request, sent over TLS. The Shared Secret Response provides the client with a 

one-time username and password. This password is then used to authenticate the 

Allocate message sent by the client to the TURN server to ask for a public IP 

address and port. For incoming data, the TURN server then stores the remote 

address and port where the data came from and forwards the data to the client. 

The TURN server is responsible for guaranteeing that packets sent to the public 

IP address route to the TURN server. TURN also allows a client to request an 

odd or even port when one is allocated, and for it to pre-allocate the next higher 

port as is useful, e.g. for the H.323 or RTP protocol. 

Interactive Connectivity Establishment (ICE) 

Interactive Connectivity Establishment (ICE) is an IETF draft, that has also been 

submitted by J.Rosenberg (dynamicsoft), R. Mahy (Cisco), C. Huitema

(Microsoft). ICE describes a methodology for NAT-Traversal for the SIP 

protocol. ICE is not a new protocol, but makes use of existing protocols, such as 

STUN, TURN and Real Specific IP (RSIP). ICE works through the mutual 

cooperation of both endpoints in a SIP dialog. ICE does not require extensions 

from STUN, TURN or RSIP. However, it does require some additional SDP 

attributes. 

Universal Plug and Play (UPnP) 

Another solution originally designed for the home market is Universal Plug and 

Play. In this scenario, the NAT is upgraded to support the UPnP protocol, and the 

client can query the NAT directly as to its external IP Address and Port number. 

However, UPnP does not scale to cascaded NATs, and there are potentially 

serious security issues with this solution, including vulnerability to denial of 

service attacks. For more information, refer to www.upnp.org 

7.6 Virtual Private Networks and Firewalls 

The next chapter deals with the integration of VPN tunneling, and specifically 

IPsec tunneling into the VOIP system. VPNs alleviate many of the problem

issues set forth in this chapter by tunneling straight through firewalls. Hence, 

many of the firewall specific issues mentioned here become moot. However, this 

“inelegant” method has some drawbacks. First, tunneling all VOIP traffic over 

VPNs prohibits firewalls from investigating incoming and outgoing packets for 

malicious traffic. Also, the centralization of security at the firewall is virtually 

lost. Also, VPN tunneling with IPsec can be incompatible with NAT. Finally, a 

host of new QoS and security issues are introduced by the integration of IPsec 

into VOIP. The next chapter covers these issues in greater detail. 

62 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

8 Encryption & IPsec 

Thus far we have focused primarily on security of the network, protecting 

endpoints, gateways, and other components, from malicious attacks. Firewalls, 

gateways, and other such devices can help keep intruders from compromising a 

network, but firewalls are no defense against an internal hacker. Another layer of 

defense is necessary at the protocol level to protect the data itself. In VOIP, as in 

data networks, this can be accomplished by encrypting the packets at the IP level 

using IPsec. This way if anyone on the network, authorized or not, intercepts 

VOIP traffic not intended for them (for instance via a packet sniffer), these 

packets will be unintelligible. The IPsec suite of security protocols and 

encryption algorithms is the standard method for securing packets against 

unauthorized viewers over data networks and will be supported by the protocol 

stack in IPv6. Hence, it is both logical and practical to extend IPsec to VOIP, 

encrypting the signal and voice packets on one end and decrypting them only 

when needed by their intended recipient. But the nature of the signaling protocols 

and the VOIP network itself prevent such a simple scheme from being used, as it 

becomes necessary for routers, proxies, etc. to read the VOIP packets. Also, 

several factors, including the expansion of packet size, ciphering latency, and a 

lack of QoS urgency in the cryptographic engine itself can cause an excessive 

amount of latency in the VOIP packet delivery. This leads to degraded voice 

quality, so once again there is a tradeoff between security and voice quality, and a 

need for speed. Fortunately, the difficulties are not insurmountable. NISTsponsored testing [29] has shown that IPsec can be incorporated into a SIP 

network with roughly a three-second additional delay in call setup times, an 

acceptable delay for many applications. This section explains the issues involved 

in successfully incorporating IPsec encryption into VOIP services. 

8.1 IPsec 

IPsec is the preferred form of VPN tunneling across the Internet. There are two 

basic protocols defined in IPsec: Encapsulating Security Payload (ESP) and 

Authentication Header (AH) (see Figure 11). Both schemes provide 

connectionless integrity, source authentication, and an anti-replay service [30]. 

The tradeoff between ESP and AH is the increased latency in the encryption and 

decryption of data in ESP and a “narrower” authentication in ESP, which 

normally does not protect the IP header “outside” the ESP header [30], although 

IKE can be used to negotiate the security association (SA), which includes the 

secret symmetric keys. In this case, the addresses in the header (transport mode) 

or new/outer header (tunnel mode) are indirectly protected, since only the entity 

that negotiated the SA can encrypt/decrypt or authenticate the packets.. Both 

schemes insert an IPsec header (and optionally other data) into the packet for 

purposes, such as authentication. 

IPsec also supports two modes of delivery: Transport and Tunnel. Transport 

mode encrypts the payload (data) and upper layer headers in the IP packet. The 

63 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

IP header and the new IPsec header are left in plain sight. So if an attacker were 

to intercept an IPsec packet in transport mode, they could not determine what it 

contained; but they could tell where it was headed, allowing rudimentary traffic 

analysis. On a network entirely devoted to VOIP, this would equate to logging 

which parties were calling each other, when, and for how long. Tunnel mode 

encrypts the entire IP datagram and places it in a new IP Packet. Both the 

payload and the IP header are encrypted. The IPsec header and the new IP

Header for this encapsulating packet are the only information left in the clear. 

Usually each “tunnel” is between two network elements such as a router or a 

gateway. In some cases, such as for mobile users, the tunnel could be between a 

router/gateway on one end and a client on the other end. The IP addresses of 

these nodes are used as the unencrypted IP address at each hop. Hence, at no 

point is a plain IP header sent out containing both the source and destination IP. 

Thus if an attacker were to intercept such packets, they would be unable to 

discern the packet contents or the origin and destination. Note that some traffic 

analysis is possible even in tunnel mode, because gateway addresses are readable. 

If a gateway is used exclusively by a particular organization, an attacker can 

determine the identity of one or both communicating organizations from the 

gateway addresses. IPsec allows nodes in the network to negotiate not only a 

security policy, which defines the security protocol and transport mode as 

described previously, but also a security association defining the encryption 

algorithm and algorithm key to be used 

Figure 11. IPsec Tunnel and Transport Modes 

64 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

8.2 The Role of IPsec inVOIP 

The prevalence and ease of packet sniffing and other techniques for capturing 

packets on an IP based network makes encryption a necessity for VOIP. Security 

in VOIP is concerned both with protecting what a person says as well as to whom

the person is speaking. IPsec can be used to achieve both of these goals as long as 

it is applied with ESP using the tunnel method. This secures the identities of both 

the endpoints and protects the voice data from prohibited users once packets leave 

the corporate intranet. The incorporation of IPsec into IPv6 will increase the 

availability of encryption, although there are other ways to secure this data at the 

application level. VOIPsec (VOIP using IPsec) helps reduce the threat of man in 

the middle attacks, packet sniffers, and many types of voice traffic analysis. 

Combined with the firewall implementations in the previous chapter, IPsec makes 

VOIP more secure than a standard phone line, where people generally assume the 

need for physical access to tap a phone line is deterrent enough. It is important to 

note, however, that IPsec is not always a good fit for some applications, so some

protocols will continue to rely on their own security features. 

8.3 LocalVPN Tunnels 

Virtual Private Networks (VPNs) are “tunnels” between two endpoints that allow 

for data to be securely transmitted between the nodes. The IPsec ESP tunnel is a 

specific kind of VPN used to traverse a public domain (the Internet) in a private 

manner. Many implementations of VOIP have attempted to make use of other 

VPN techniques, including VPN tunneling within an organization’s’s intranet. 

The use and benefits of VPNs in IPsec have been great enough for some to claim

“VOIP is the killer app for VPNs” [26]. VPN tunnels within a corporate LAN or 

WAN are much more secure and generally faster than the IPsec VPNs across the 

Internet because data never traverses the public domain, but they are not 

scaleable. This sort of implementation has a physical limit at the size of the 

private network, and as VOIP becomes more widely spread, it is not practical for 

an implementation to regard calls outside the local network as a black box. Also, 

no matter how the VPN is set up, the same types of attacks and issues associated 

with IPsec VPNs are applicable, so we consider here only the case of IPsec 

tunneling and assume the security solutions can be scaled down to an internal 

network if needed. 

8.4 Difficulties Arising from VOIPsec 

IPsec has been included in IPv6. It is a reliable, robust, and widely implemented 

method of protecting data and authenticating the sender. However, there are 

several issues associated with VOIP that are not applicable to normal data traffic. 

Of particular interest are the Quality of Service (QoS) issues discussed in Chapter 

3. Chief among these are latency, jitter, and packet loss. These issues are 

introduced into the VOIP environment because it is a real time media transfer, 

with only 150 ms to deliver each packet. In standard data transfer over TCP, if a 

packet is lost, it can be resent by request. In VOIP, there is no time to do this. 

65 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

Packets must arrive at their destination and they must arrive fast. Of course the 

packets must also be secure during their travels, thus the introduction of VOIPsec. 

However, the price of this security is a decisive drop in QoS caused by a number 

of factors. 

A 2002 study by researchers at the University of Milan [8] focused on the effect 

of VOIPsec on various QoS issues and on the use of header compression as a 

solution to these problems. They studied several codecs, encryption algorithms, 

and traffic patterns to garner a broad description of these effects. Their results are 

an integral part of the next few sections. 

Some empirical results developed by Cisco are available as well [31] 

− Delay 

o Processing—PCM to G.729 to packet 

o Encryption — ESP encapsulation + 3DES 

o Serialization — time it takes to get a packet out of the router, each 

“hop” generally has fixed delay. 

 IPsec overhead: about 40 bytes (depending configuration) 

 IP header: 20 bytes 

 UDP + RTP headers: 20 bytes 

 RTP header compression: 3 bytes for IP+UDP+RTP 

− Effects on 8 kbps CODEC (voice data: 20 bytes) 

o clear text voice has an overhead of 3 bytes, which suggests required 

bandwidth of approximately 9 kbps 

o IPsec encrypted voice: overhead 80 bytes and required bandwidth 40 

kbps 

8.5 Encryption / Decryption Latency

The studies performed by Barbieri et al. revealed the cryptographic engine as a 

bottleneck for voice traffic transmitted over IPsec. The driving factor in the 

degraded performance produced by the cryptography was the scheduling 

algorithms in the crypto-engine itself, which will be covered in Section 8.6. 

However, there still was significant latency due to the actual encryption and 

decryption. Barbieri et al. set up a controlled experiment to measure the effect of 

encryption and decryption on throughput. They tested four cryptographic 

algorithms on a fully VOIP dedicated network with a 100Mbps link (to negate 

saturation issues) using the same traffic in plain form as a benchmark. The 

algorithms tested were (in increasing order of computational expense) DES, 

3DES, NULL (no encryption) + SHA-1, and 3DES + SHA-1. The results showed 

that the computationally lighter algorithms achieved better throughput than the 

more expensive ones. The disparities between each of the algorithms represent 

the relative latencies associated with the computational time for each algorithm. 

The range in throughput is significant, with a difference of approximately 500 

packets per second between DES and 3DES + SHA-1 at a high traffic volume. 

66 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

Encryption/decryption latency is a problem for any cryptographic protocol, 

because much of it results from the computation time required by the underlying 

encryption. With VOIP’s use of small packets at a fast rate and intolerance for 

packet loss, maximizing throughput is critical. However 

, this comes with a price, because although DES is the fastest of these encryption 

algorithms, it is also the easiest to crack. Current rules prohibit the use of DES for 

protection of US Government information. Thus, designers are once again forced 

to toe the line between security and voice quality. Two solutions to this problem

are using faster encryption algorithms (9.3) and incorporating QoS into the 

crypto-engine (9.4). Latency is less of a problem for management and/or 

signaling data than for voice channel traffic.

 

8.6 Scheduling and the Lackof QoS in the Crypto-Engine

The crypto-engine is a severe bottleneck in the VOIP network. As just noted, the 

encryption process has a debilitating effect on QoS, but this is not the highest 

degree factor in the slowdown. Instead, the driving force behind the latency 

associated with the crypto-engine is the scheduling algorithm for packets that 

entered the encryption/decryption process. While routers and firewalls take 

advantage of QoS to determine priorities for packets, crypto-engines provide no 

support for manual manipulation of the scheduling criteria. In ordinary data 

traffic this is less of an issue because inordinately more packets pass through the 

router than the crypto-engine, and time is not as essential. But in VOIP, a 

voluminous number of small packets must pass through both the crypto engine 

and the router. Considering the time urgency issues of VOIP, the standard FIFO 

scheduling algorithm employed in today’s crypto-engines creates a severe QoS 

issue. 

Barbieri et al. found that the throughput of the crypto scheduling algorithm 

actually increased with larger packet sizes. They concluded that scheduling a 

greater number of packets had a more degrading effect on performance than 

encrypting/decrypting fewer (but larger) packets. So with the numerous small 

packets VOIP uses, the crypto engine soon reaches the saturation point, and 

throughput is compromised. This accounts for the asymptotic behavior of the 

throughput for encrypted traffic against plain traffic’s continuous rise and the 

increased delay of encrypted traffic that Barbieri et al. observed. 

These QoS violations derived from the crypto-engine are exacerbated by the 

presence of actual data traffic on a VOIP network. Since one of the primary 

motivations for the development of VOIP is the ability of voice and data to share 

the same network, this scenario is to be expected. Barbieri et al.’s experiments 

showed that such a combination of traffic has disastrous effects on VOIPsec. This 

is especially true if the heterogeneous data needs to be encrypted and decrypted. 

Since the crypto-engine has no functionality for changing its own priority schema

based on the type of traffic it is presented with, the VOIP packets are at the mercy 

of the FIFO scheduling algorithm, and are often left waiting behind larger 

67 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

heterogeneous packets, despite the lesser urgency of these large packets. The 

non-uniform pattern of data traffic also contributes to a great deal of jitter in 

VOIP. The variation in the bandwidth usage caused by heterogeneous packets 

wreaks havoc with the delay times of the fairly uniform VOIP packets, causing 

them to arrive in spurts. If these spurts exceed the timing mechanism associated 

with the buffer, then packet loss can occur. The development of VOIP-aware 

crypto schedulers would help to relieve this problem. 

8.7 ExpandedPacket Size 

IPsec also increases the size of packets in VOIP, which leads to more QoS issues. 

It has been shown that increased packet size increases throughput through the 

crypto-engine, but to conclude from this that increased packet size due to IPsec 

leads to better throughput would be fallacious. The difference is that the increase 

in packet size due to IPsec does not result in an increased payload capacity. The 

increase is actually just an increase in the header size due to the encryption and 

encapsulation of the old IP header and the introduction of the new IP header and 

encryption information. (see 8.1). This leads to several complications when 

IPsec is applied to VOIP. First, the effective bandwidth is decreased as much as 

63% [9]. Thus connections to single users in low bandwidth areas (i.e. via 

modem) may become infeasible. The bandwidth performance reductions for 

various encryption algorithms are presented in [8]. The size discrepancy can also 

cause latency and jitter issues as packets are delayed by decreased network 

throughput or bottlenecked at hub nodes on the network (such as routers or 

firewalls). 

8.8 IPsec and NAT Incompatibility

IPsec and NAT compatibility is far from ideal. NAT traversal completely invalidates 

the purpose of AH because the source address of the machine behind the NAT is 

masked from the outside world. Thus, there is no way to authenticate the true sender 

of the data. The same reasoning demonstrates the inoperability of source 

authentication in ESP. We have defined this as an essential feature of VOIPsec, so 

this is a serious problem. There are several other issues that arise when ESP traffic 

attempts to cross a NAT. If only one of the endpoints is behind a NAT, the situation 

is easier [32] If both are behind NATs, IKE negotiation can be used for NAT 

traversal, with UDP encapsulation of the IPsec packets.

68 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

9 Solutions to the VOIPsec Issues 

Thus far, we have raised a number of significant concerns with IPsec’s role in 

VOIP. However, many of these technical problems are solvable. Despite the 

difficulty associated with these solutions it is well worth the establishment of a 

secure implementation of VOIPsec. 

9.1 Encryption at theEnd Points

One proposed solution to the bottlenecking at the routers due to the encryption 

issues is to handle encryption/decryption solely at the endpoints in the VOIP 

network [33]. One consideration with this method is that the endpoints must be 

computationally powerful enough to handle the encryption mechanism. But 

typically endpoints are less powerful than gateways, which can leverage hardware 

acceleration across multiple clients. Though ideally encryption should be 

maintained at every hop in a VOIP packet’s lifetime, this may not be feasible with 

simple IP phones with little in the way of software or computational power. In 

such cases, it may be preferable for the data be encrypted between the endpoint 

and the router (or vice versa) but unencrypted traffic on the LAN is slightly less 

damaging than unencrypted traffic across the Internet. Fortunately, the increased 

processing power of newer phones is making endpoint encryption less of an issue. 

In addition, SRTP and MIKEY are future protocols for media encryption and key 

management enabling secure interworking between H.323 and SIP based clients. 

9.2 Secure Real Time Protocol (SRTP) 

RTP (Real-time Transport Protocol) is commonly used for the transmission of 

real-time audio/video data in Internet telephony applications. Without protection 

RTP is considered insecure, as a telephone conversation over IP can easily be 

eavesdropped. Additionally, manipulation and replay of RTP data could lead to 

poor voice quality due to jamming of the audio/video stream. Modified RTCP 

(Real-time Transport Control Protocol) data could even lead to an unauthorized 

change of negotiated quality of service and disrupt the processing of the RTP 

stream. 

The Secure Real-time Protocol is a profile of the Real-time Transport Protocol 

(RTP) offering not only confidentiality, but also message authentication, and 

replay protection for the RTP traffic as well as RTCP (Real-time Transport 

Control Protocol). SRTP was being standardized at the IETF in the AVT working 

group. It was released as RFC 3711 in March 2004. 

 

SRTP provides a framework for encryption and message authentication of RTP 

and RTCP streams. SRTP can achieve high throughput and low packet expansion. 

SRTP is independent of a specific RTP stack implementation and of a specific 

key management standard, but Multimedia Internet Keying (MIKEY) has been 

designed to work with SRTP. 

69 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

AES in counter mode is the default algorithm, if encryption is desired. AES-f8 

mode is an option for UMTS applications. The pre-defined authentication 

transform is HMAC-SHA1. The default session authentication key-length is 160 

bits, the default authentication tag length is 80 bits. The key derivation function is 

AES in counter mode with a 128-bit master key from the key management. 

Interface for hardware-crypto support (e.g. IP phones). In comparison to the 

security options for RTP there are some advantages to using SRTP. The 

advantages over the RTP standard security and also over the H.235 security for 

media stream data are listed below. 

SRTP provides increased security, achieved by 

• Confidentiality for RTP as well as for RTCP by encryption of the 

respective payloads; 

• Integrity for the entire RTP and RTCP packets, together with replay 

protection; 

• The possibility to refresh the session keys periodically, which limits the 

amount of cipher text produced by a fixed key, available for an adversary 

to cryptanalyze; 

• An extensible framework that permits upgrading with new cryptographic 

algorithms; 

• A secure session key derivation with a pseudo-random function at both 

ends; 

• The usage of salting keys to protect against pre-computation attacks; 

• Security for unicast and multicast RTP applications. 

SRTP has improved performance attained by 

• Low computational cost asserted by pre-defined algorithms; 

• Low bandwidth cost and a high throughput by limited packet expansion 

and by a framework preserving RTP header compression efficiency; 

• Small footprint that is a small code size and data memory for keying 

information and replay lists. 

The following characteristics also argue for SRTP: 

• It is defined as a profile of RTP, so that it can be easily integrated into 

existing RTP stacks. For example SRTP may use RTP padding because 

the encrypted portion is the exact size of the plaintext for the pre-defined 

algorithms. 

• It provides independence from the underlying transport, network, and 

physical layers used by RTP, in particular high tolerance to packet loss 

and re-ordering, and robustness to transmission bit-errors in the encrypted 

payload. 

• It lightens the burden of the key management due to the fact that a single 

master key can provide keying material for confidentiality and integrity 

protection, both for the SRTP stream and the corresponding SRTCP 

stream. For special requirements a single master key can protect several 

70 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

SRTP streams. 

Because SRTP is defined as an RTP profile it may be used with existing 

multimedia standards. H.323 SRTP support is defined within the H.235 Annex G 

(currently in draft status), for SIP or more precisely SDP enhancements have been 

defined to transport the key management data necessary for SRTP. Thus, the 

combination of SRTP and MIKEY may be used to provide end-to-end encryption 

even between different multimedia signaling standards like H.323 and SIP. 

9.3 Key Management for SRTP – MIKEY 

SRTP uses a set of negotiated parameters from which session keys for encryption, 

authentication and integrity protection are derived. MIKEY (Section 4.2.2.2) 

describes a key management scheme that addresses real-time multimedia 

scenarios (e.g. SIP calls and RTSP sessions, streaming, unicast, groups, multicast) 

and is currently being standardized within the IETF’s MSEC group. The focus 

lies on the setup of a security association for secure multimedia sessions including 

key management and update, security policy data, etc., such that requirements in a 

heterogeneous environment are fulfilled. MIKEY also supports the negotiation of 

single and multiple crypto sessions. This is especially useful for the case where 

the key management is applied to SRTP, since here RTP and RTCP may to be 

secured independently. Deployment scenarios for MIKEY comprise peer-to-peer, 

simple one-to-many, and small-size interactive group scenarios. 

MIKEY supports the negotiation of cryptographic keys and security parameters 

(SP) for one or more security protocols. This results in the concept of crypto 

session bundles, which describe a collection of crypto sessions that may have a 

common Traffic Encryption Key (TEK) Generation Key (TGK) and belonging 

session security parameters. 

MIKEY has some important properties: 

o MIKEY can be implemented as an independent software library to be easily 

integrated in a multimedia communication protocol. It offers independency of 

a specific communication protocol (SIP, H.323, etc.) 

o Establishment of key material within a 2-way handshake, therefore best suited 

for real-time multimedia scenarios 

o There are four options for Key Distribution: 

o Preshared-key 

o Public-key encryption 

o Diffie-Hellman key exchange protected by public-key encryption 

o Diffie-Hellman key exchange protected with preshared-key and 

keyed hash functions (using an MIKEY extension (DHHMAC)) 

o Re-keying Support 

o Multicast Support (one sender) 

71 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

9.4 Better Scheduling Schemes 

The incorporation of AES or some other speedy encryption algorithm could help 

temporarily alleviate the bottleneck, but this is not a scalable solution because it 

does not address the highest degree cause of the slowdown. Without a way for 

the crypto-engine to prioritize packets, the engine will still be susceptible to DoS 

attacks and starvation from data traffic impeding the time-urgent VOIP traffic. A 

few large packets can clog the queue long enough to make the VOIP packets over 

150 ms late (sometimes called head-of-line blocking), effectively destroying the 

call. Ideally, the crypto-engine would implement QoS scheduling to favor the 

voice packets, but this is not a realistic scenario due to speed and compactness 

constraints on the crypto-engine. One solution implemented in the latest routers 

is to schedule the packets with QoS in mind prior to the encryption phase. 

Although this heuristic solves the problem for all packet poised to enter the crypto 

engine at a given time, it does not address the problem of VOIP packets arriving 

at a crypto–engine queue that is already saturated with previously scheduled data 

packets. QoS prioritizing can also be done after the encryption process provided 

your encryption procedures preserve the ToS bits from the original IP header in 

the new IPsec header. This functionality is not guaranteed and is dependent on 

one’s network hardware and software, but if it is implemented it allows for QoS 

scheduling to be used at every hop the encrypted packets encounter. There are 

security concerns any time information on the contents of a packet is left in the 

clear, including this ToS-forwarding scheme, but with the sending and receiving 

addresses concealed, this is not as egregious as a cursory glance would make it 

seem. Still neither the pre-encryption or post-encryption schemes actually 

implement QoS or any other prioritizing scheme to enhance the crypto-engine’s 

FIFO scheduler. Speed and compactness constraints on this device may not allow 

such algorithms to be applied for some time. 

9.5 Compressionof Packet Size

A novel approach to the QoS issues associated with VOIPsec is proposed by 

Barbieri et al. at the conclusion of their studies of VOIPsec traffic. Their solution 

targets the increase in packet size stemming from the use of IPsec. They 

implemented cIPsec: a version of IPsec that compresses the internal header of a 

packet down to approximately four bytes. This is possible because much of the 

data in the internal headers of a packet remained constant or was duplicated in the 

outer header. 

The initial test results reported from the University of Milan indicate that the 

compression of IPsec headers results in bandwidth usage comparable to that of 

plain IP. This in turn results in considerably less jitter, latency, and better cryptoengine performance. The crypto-engine performance also improves. There is, 

of course, a price for these speedups. The compression scheme puts more strain 

on the CPU and memory capabilities of the endpoints in order to achieve the 

72 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

compression, and, of course, both ends of a connection must use the same

compression algorithm. However, the study found that the time lost to 

compression was made up for at the encryption phase, as the crypto-engine is 

more efficient with the compressed packets. One thing they did not consider is 

the tremendous strain put on end-point CPU’s as opposed to the crypto-engine. 

The endpoint CPU may be computationally slow (in the case of a simple VOIP 

phone) or may be performing many more operations than just VOIP (in the case 

of a PC-based phone). In either case, the actual time required to perform the 

compression may take much longer than the time saved in the crypto-engine. It 

remains to be seen if this is the case, as Barbieri’s model of cIPsec was not tested 

under high CPU load conditions.

It is important to note that the compression scheme used in cIPsec only 

compresses the packet header information. The compression QoS issues 

associated with audio codecs are not applicable in this scenario because no actual 

media is being condensed, only the IP headers. However, packet loss does have 

an exacerbated detrimental effect (for a different reason) on packets compressed 

under the cIPsec scheme. Barbieri’s scheme needs to maintain information at the 

system endpoints regarding the current session. When packets are lost, they 

cannot be re-sent and the endpoints need to resynchronize [9]. However, the time

saved in the crypto-engine and the security provided may be well worth this price 

of this approach. Further testing will be required to determine the validity of this 

solution under diverse network and environmental conditions. 

9.6 Resolving NAT/IPsec Incompatibilities

There are solutions to the IPsec/NAT incompatibility problem previously outlined 

in section 8.2. Straka [32] discusses several of these, including Realm-Specific IP 

RSIP), IPv6 Tunnel Broker, IP Next Layer (IPNL), and UDP encapsulation. 

RSIP is designed as a replacement for NAT and provides a clear tunnel between 

hosts and the RSIP Gateway. RSIP supports both AH and ESP, but implementing 

RSIP would require a significant overhaul of the current LAN architecture so 

while it is quite an elegant solution, it is currently infeasible. Perhaps as a result 

of these problems, RSIP is not widely used. The IPv6 tunnel broker method uses 

an IPv6 tunnel as an IPsec tunnel, and encapsulates an IPv6 packet in an IPv4 

packet. But this solution also requires LAN upgrades and doesn’t work in 

situations where multiple NATs are used. IPNL introduces a new layer into the 

network protocols between IP and TCP/UDP to solve the problem, but IPNL is in 

competition with IPv6 and IPv6 is a much more widely used standard. 

The most likely widespread solution to the problem of NAT traversal is UDP 

encapsulation of IPsec. This implementation is supported by the IETF and 

effectively allows all ESP traffic to traverse the NAT. In tunnel mode, this model 

wraps the encrypted IPsec packet in a UDP packet with a new IP header and a 

new UDP header, usually using port 500. This port was chosen because it is 

currently used by IKE peers to communicate so overloading the port does not 

require any new holes to be punched in the firewall [34]. The SPI field within the 

73 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

UDP-encapsulated packet is set to zero to differentiate it from an actual IKE 

communication. This solution allows IPsec packets to traverse standard NATs in 

both directions. The adoption of this standard method should allow VOIPsec 

traffic to traverse NATs cleanly, although some extra overhead is added in the 

encapsulation/decapsulation process. IKE negotiation will also be required to 

allow for NAT traversal. The problem still remains that IP-based authentication of

the packets cannot be assured across the NAT, (although fully qualified domain 

names could be used) but the use of a shared secret (symmetric key) negotiated 

through IKE could provide authentication. It is important to note that IP-based 

authentication is weak compared with methods using cryptographic protocols. 

74 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

10 Planning for VOIP Deployment 

VOIP is still an emerging technology, so it is somewhat speculative to develop a 

complete picture of what a mature worldwide VOIP network will one day look 

like. As the emergence of SIP has shown, new technologies and new protocol 

designs have the ability to change VOIP. The situation is analogous to the state of 

the Internet in the late 80s and early 90s. Competing protocols and designs for the 

infrastructure of the net flourished at the time, but as the purpose of the Internet 

became more defined with the emergence of the world wide web and other staples 

of today’s net, the structure and protocols became standardized and 

interoperability became much easier. The same may one day be true of VOIP. 

Although there are currently many different architectures and protocols to choose 

from, eventually a dominant standard will emerge. 

The most obvious of these competing standards are SIP and H.323. Comparisons 

are made in numerous academic papers including [16, 18, and 15]. SIP is a fast 

growing protocol with similarities to current Internet standards such as HTTP, but 

it has yet to reach the level of deployment of H.323 [15]. The opinion of many 

academics [16, 18] seems to favor SIP, and we have seen that some of the security 

issues associated with VOIP become simpler with the SIP scheme. It is not clear 

which standard will prevail in the market. It is misleading to portray the choice 

between SIP and H.323 as mutually exclusive. In fact, in today’s nonstandardized VOIP environment, organizations looking to integrate several VOIP 

networks ought to support both protocols. Several companies have developed 

infrastructure elements to enable multi-protocol telephony. As voice and data 

networks converge, support for both protocols is essential for a robust and 

forward-looking network. Although the future will probably see the emergence of 

one of these protocols as the defined standard in the field the present 

disorganization makes support for both protocols in a VOIP network a pertinent 

issue. Deploying a VOIP network in today’s non-standardized world requires 

support for both protocols. Consequently, organizations moving to VOIP should 

seek out gateways and other network elements that can support both H.323 and 

SIP. Such a strategy helps to ensure a stable and robust VOIP network in the 

years that come, no matter which protocol prevails. 

The other high-level issue in VOIP security today is the choice of end-to-end 

VPNs versus firewall-based VPNs. That is, VOIP traffic must traverse firewalls 

one way or the other. The question becomes, should one build firewalls with 

ALGs, proxies, firewall control proxies, and IPsec functionality to facilitate this, 

or simply tunnel all VOIP traffic straight through the firewall with a VPN. The 

security benefits and administrative troubles associated with each of these 

implementations have been presented in detail in this publication. The use of 

VPNs has been touted by many industry articles as the definitive solution to the 

tribulations posed by firewall and NAT traversal in tunnel mode. However, 

much of their research has focused on small-scale operations where VOIP phones 

75 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

are not used in the volume needed to overwhelm the crypto-engines or congest the 

network enough to cause a significant downturn in QoS. Thus it is not clear that 

the large-scale implementation of a VPN tunneling system for VOIP is an 

effective solution for an enterprise network [see Broadcom in 27], as there is 

significant overhead associated with VOIPsec. Also, the ability to use firewalls 

for analyzing VOIP traffic for malicious or suspicious patterns would be lost. 

This being said, the implementation of VOIP aware firewalls and proxies incurs a 

significant cost now and in the future. Such protocol specific hardware would 

need to be upgraded each time standards evolve. A third solution that has not 

been fully developed yet is a hybrid system, where call setup information (H.323 

or SIP) is sent through a VOIP aware gateway/firewall but the RTP traffic itself is 

encrypted and tunneled over VPN. The call setup protocols could be secured 

using their proprietary authentication mechanisms [17] in place of the IPsec 

tunnel. This would seemingly combine the network protection of the firewall and 

the data security/protection of IPsec. Also, the robust authentication mechanisms

and abstraction of the voice network from the data network accomplished by SIP 

proxies and H.323 gateways/gatekeepers would be preserved. However, no 

expanded study has been done on the ramifications of this hybrid approach. 

There are a number of privacy issues regarding storage of call detail records. 

Agencies and other organizations should review these issues with their legal 

advisors. US Government guidance on these issues includes the following: 

• the Privacy Act of 1974. 

• Office of Management and Budget "Guidance on the Privacy Act Implications of 

Call Detail Programs to Manage Employees' Use of the Government's 

Telecommunication System" (See FEDERAL REGISTER, 52 FR 12990, April 20, 

1987). 

• NARA General Records Schedule 12, which requires a 36-month retention of

telephone CDR records 

http://www.archives.gov/records_management/ardor/grs12.html

• 21 CFR 102-172, Federal Management Regulation (FMR), 

Telecommunications Management Policy. 21 CFR 102-172 . Note that 21 

CFR 102-172 replaces 21 CFR 101-35, Federal Property Management 

Regulation (FPMR), Telecommunications Management Policy, which expired 

in August 2001. 

The construction of a VOIP network is an intricate procedure that should be 

studied in great detail before being attempted. New risks can be introduced, and 

vulnerabilities of data packet networks appear in new guises for VOIP (see 

Appendix A for more detailed discussion of vulnerabilities of VOIP and their 

relation to data network vulnerabilities). The integration of a VOIP system into an 

already congested or overburdened network could be catastrophic for an 

76 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

organization’s technology infrastructure. There is no easy “one size fits all” 

solution to the issues discussed in these chapters. The use of VPNs, vs. ALG-like 

solutions and the choice of SIP or H.323 are decisions that must be made based on 

the specific nature of the current network and the VOIP network to be installed. 

VOIP can be done securely, but the path is not smooth. It will likely be several 

years before standards issues are settled and VOIP systems become a mainstream

commodity. Until then, organizations must proceed cautiously, and not assume

that VOIP components are just more peripherals for the local network. Above all, 

it is important to keep in mind the unique requirements of VOIP, acquiring the 

right hardware and software to meet the challenges of VOIP security. 

77 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

REFERENCES

 

1 F. Robles. “The VOIP Dilemma”, SANS Institute, 

http://www.sans.org/rr/whitepapers/voip/1452.php

2. National Institute of Standards and Technology, DRAFT FIPS Publication 199, 

Standards for Security Categorization of Federal Information and Information 

Systems, September 18, 2003 http://csrc.nist.gov/publications/drafts/draft-fipspub-199.pdf 

3. W.C. Hardy, QoS Measurement and Evaluation of Teleocmmunication Quality 

of Service, John Wiley & Sons, 2001. 

4. W.C. Hardy, VOIP Service Quality: Measuring and Evaluating PacketSwitched Voice, McGraw-Hill, 2003. 

5. International Telecommunications Union. ITU-T Recommendation G.114 

(1998): "Delay". 

6. P. Mehta and S. Udani, “Overview of Voice over IP”. Technical Report MSCIS-01-31, Department of Computer Information Science, University of 

Pennsylvania, February 2001. 

7. B. Goode, “Voice Over Internet Protocol (VOIP)”. Proceedings of thee IEEE, 

VOL. 90, NO. 9, Sept. 2002. 

8. R. Barbieri, D. Bruschi, E Rosti, “Voice over IPsec: Analysis and Solutions”. 

Proceedings of the 18th Annual Computer Security Applications Conference,2002. 

9. C-N. Chuah, “Providing End-to-End QoS for IP based Latency sensitive 

Applications.”. Technical Report, Dept. of Electrical Engineering and Computer 

Science, University of California at Berkeley, 2000. 

10. B. Goode, “Voice Over Internet Protocol (VOIP)”. Proceedings of thee IEEE, 

VOL. 90, NO. 9, Sept. 2002. 

11. Anonymous, “Voice Over IP Via Virtual Private Networks: An Overview”. 

White Paper, AVAYA Communication, Feb. 2001. 

12. R. Sinden, “Comparison of Voice over IP with circuit switching techniques”. 

Department of electronics and Computer Science, Southampton University, UK, 

Jan. 2002. 

13. K. Percy and M. Hommer, “Tips from the trenches on VOIP”. Network World 

Fusion, Jan. 2003 

78 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

 

14. J. Larson, T. Dawson, M. Evans, J.C. Straley, “Defending VoIP Networks 

from DDoS Attacks”, GlobeCom 2004 VoIP Security Workshop. 

 

15. Anonymous, “H.323 and SIP Integration”. White Paper, Cisco Systems, 2001. 

16. K. Siddiqui, M. Kamran, S. Tajammul, “Comparison of H.323 and SIP for IP 

Telephony Signaling”. In Proceedings of IEEE 4th International Multioptics 

Conference, Lahore, Pakistan, Dec. 2001. 

17. J. Thalhammer, “Security in VOIP-Telephony Systems”. Master Thesis, 

Institute for Applied Information Processing and Communications, Graz U. of 

Technology, 

18. H. Schulzrinne and J. Rosenberg. A comparison of SIP and H.323 for Internet 

telephony. In Proc. International Workshopon Network and Operating System 

Support for Digital Audio and Video (NOSSDAV), Cambridge, England, 

July1998. 

19. M. Marjalaakso, “Security Requirements and Constraints of VOIP”. 

Department of Eleectrical Engineering and Telecommunications, Helsinki

University of Technology, 2001. 

20 Jablon, “Strong Password-Only Authenticated Key Exchange”, Mar. 1997, 

http://www.integritysciences.com/speke97.html

21. Anonymous, “H.323 and firewalls: The problems and Pitfalls of Getting 

H.323 safely through firewalls” Developer note, Intel Corporation, Apr. 1997. 

22. J. Rosenberg and H. Schulzrinne, “SIP Traversal through Residential and 

Enterprise NATs and Firewalls”. Internet Draft, Internet Engineering Task Force, 

Mar. 2001. 

23. J. Halpern, “IP Telephony Security in Depth”. White Paper, Cysco Systems, 

2002. 

24. Anonymous, “Voice Over IP Via Virtual Private Networks: An Overview”. 

White Paper, AVAYA Communication, Feb. 2001. 

25. A. Romeo, G. Romolotti, M. Mattavelli, D. Mlynek, “Cryptosystem

Architectures for very High Throughput Multimedia Encryption: The RPK 

Solution”, Swiss Federal Institute of technology, Lausanne, Switzerland, 1999. 

26. A. Conry-Murray, “Emerging Technology: Security and Voice over IP – Let’s 

Talk”. Commweb, Nov. 2002. 

79 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

 

27. P. Hochmuth and T. Greene, “Firewall limits vex VOIP users”. Network 

World Fusion, July 2002. 

28. Anonymous, “Traversing Firewalls and NATs With Voice and Video Over 

IP: An Examination of the Firewall/NAT Problem, Traversal Methods, and their 

Pros and Cons” . Wainhouse Research, Apr. 2002. 

29. Telcordia Technologies, “Performance and Security Analysis of SIP using 

IPsec”, National Institute of Standards and Technology, January, 2004. 

 

30. G. Egeland, “Introduction to IPsec in IPv6”. Eurescom, 

http://www.eurescom.de/~publicwebdeliverables/P1100series/P1113/D1/pdfs/pir1

/41_IPsec_intro.pdf

31. Cisco Networkers 2000, 

http://www.cisco.com/networkers/nw00/pres/2403.pdf

32. R. Straka, “IPsec VPN Traffic and NAT / NAPT Traversal: Searching for a 

Universal Solution”. Center for Information Security, University of Tulsa, Dec. 

2002. 

33. O. Arkin, “Why E.T. Can’t Phone Home?: Security Risk Factors with IP 

Telephony based Networks” . Sys-Security Group, Nov. 2002. 

34. L. Phifer, “Slipping IPsec Past NAT”. ISP Planet, Apr. 2001. 

34. Internet Draft: End-to-middle Security in the Session Initiation Protocol 

(SIP), K.Ono, S. Tachimoto, February 2004, Work in Progress 

http://www.ietf.org/Internet-drafts/draft-ono-sipping-end2middle-security-01.txt

35. Internet Draft: A Mechanism to Secure SIP information inserted by 

Intermediaries,M.Barnes, October 2003, Work in Progress 

http://search.ietf.org/Internet-drafts/draft-barnes-sipping-sec-inserted-info-01.txt

36. Request for Comments 3489: STUN: Simple Traversal of UDP through 

Network Address Translators, , J.Rosenberg, et al, March 2003. 

http://www.ietf.org/rfc/rfc3489.txt 

80 

NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

A Appendix: VOIP Risks, Threats, and Vulnerabilities 

This appendix details some of the potential threats and vulnerabilities in a VOIP 

environment, including vulnerabilities of both VOIP phones and switches. Threat 

discussion is included because the varieties of threats faced by an organization 

determine the priorities in securing its communications equipment. Not all threats 

are present in all organizations. A commercial firm may be concerned primarily 

with toll fraud, while a government agency may need to prevent disclosure of 

sensitive information because of privacy or national security concerns. 

Information security risks can be broadly categorized into the following three 

types: confidentiality, integrity, and availability, (which can be remembered with 

the mnemonic “CIA”). Additional risks relevant to switches are fraud and risk of 

physical damage to the switch, physical network, or telephone extensions. 

Packet networks depend for their successful operation on a large number of 

configurable parameters: IP and MAC (physical) addresses of voice terminals, 

addresses of routers and firewalls, and VOIP specific software such as Call 

Managers and other programs used to place and route calls. Many of these 

network parameters are established dynamically every time a network component 

is restarted, or when a VOIP telephone is restarted or added to the network. 

Because there are so many places in a network with dynamically configurable 

parameters, intruders have a wide array of potentially vulnerable points to attack. 

Vulnerabilities described in this section are generic and may not apply to all 

systems, but investigations by NIST and other organizations have found these 

vulnerabilities in a number of VOIP systems. In addition, this list is not 

exhaustive; systems may have security weaknesses that are not included in the 

list. For each potential vulnerability, a recommendation is included to eliminate or 

reduce the risk of compromise. 

A.1 Confidentiality and Privacy 

Confidentiality refers to the need to keep information secure and private. For 

home computer users, this category includes confidential memoranda, financial 

information, and security information such as passwords. In a 

telecommunications switch, eavesdropping on conversations is an obvious 

concern, but the confidentiality of other information on the switch must be 

protected to defend against toll fraud, voice and data interception, and denial of 

service attacks. Network IP addresses, operating system type, telephone 

extension to IP address mappings, and communication protocols are all examples 

of information that, while not critical as individual pieces of data, can make an 

attacker’s job easier 

With conventional telephones, eavesdropping usually requires either physical 

access to tap a line, or penetration of a switch. Attempting physical access 

increases the intruder’s risk of being discovered, and conventional PBXs have 

81 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

fewer points of access than VOIP systems. With VOIP, opportunities for 

eavesdroppers increase dramatically, because of the many nodes in a packet 

network. 

 

Switch Default Password Vulnerability 

 

It is common for switches to have a default login/password set, e.g., 

admin/admin, or root /root. This vulnerability also allows for wiretapping 

conversations on the network with port mirroring or bridging. An attacker with 

access to the switch administrative interface can mirror all packets on one port to 

another, allowing the indirect and unnoticeable interception of all 

communications. Failing to change default passwords is one of the most 

common errors made by inexperienced users. If possible, remote access to the 

graphical user interface should be disabled to prevent the interception of 

plaintext administration sessions. Some devices provide the option of a direct 

USB connection in addition to remote access through a web browser interface. 

Disabling port mirroring on the switch should also be considered. 

Classical Wiretap Vulnerability 

Attaching a packet capture tool or protocol analyzer to the VOIP network 

segment makes it easy to intercept voice traffic. 

 A good physical security policy for the deployment environment is a general first 

step to maintaining confidentiality. Disabling the hubs on IP Phones as well as 

developing an alarm system for notifying the administrator when an IP Phone has 

been disconnected will allow for the possible detection of this kind of attack. 

ARP Cache Poisoning and ARP Floods 

Because many systems have little authentication, an intruder may be able to log onto

a computer on the VOIP network segment, and then send ARP commands corrupting 

ARP caches on sender(s) of desired traffic, then activate IP. An ARP flood attack on

the switch could render the network vulnerable to conversation eavesdropping. 

Broadcasting ARP replies blind is sufficient to corrupt many ARP caches. 

Corrupting the ARP cache makes it possible to re-route traffic to intercept voice and

data traffic. Use authentication mechanisms provided wherever possible and limit 

physical access to the VOIP network segment. 

Web Server interfaces 

Both VOIP switches and voice terminals are likely to have a web server interface 

for remote or local administration. An attacker may be able to sniff plaintext 

HTTP packets to gain confidential information. This would require access to the 

local network on which the server resides. If possible, do not use an HTTP 

82 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

server. If it is necessary to use a web server for remote administration, use the 

more secure HTTPS (HTTP over SSL or TLS) protocol. 

IP Phone Netmask Vulnerability 

A similar effect of the ARP Cache Vulnerability can be achieved by assigning a 

subnet mask and router address to the phone crafted to cause most or all of the 

packets it transmits to be sent to an attacker’s MAC address. Again, standard (1q 

aware) IP forwarding makes the intrusion all but undetectable. 

 

A firewall filtering mechanism can reduce the probability of this attack. 

Remote access to IP phones is a severe risk. 

Extension to IP Address Mapping Vulnerability 

Discovering the IP address corresponding to any extension requires only calling 

that extension and getting an answer. A protocol analyzer or packet capture tool 

attached to the hub on the dialing instrument will see packets directly from the 

target instrument once the call is answered. Knowing the IP address of a 

particular extension is not a compromise in itself, but makes it easier to 

accomplish other attacks. For example, if the attacker is able to sniff packets on 

the local network used by the switch, it will be easy to pick out packets sent and 

received by a target phone. Without knowledge of the IP address of the target 

phone, the attacker’s job may be much more difficult to accomplish and require 

much longer, possibly resulting in the attack being discovered. Disabling the 

hub on the IP Phone will prevent this kind of attack. However, it is a rather 

simple task to turn the hub back on. 

A.2 Integrity Issues 

Integrity of information means that information remains unaltered by 

unauthorized users. For example, most users want to ensure that bank account 

numbers cannot be changed by anyone else, or that passwords are changed only 

by the user or an authorized security administrator. Telecommunication switches 

must protect the integrity of their system data and configuration. Because of the 

richness of feature sets available on switches, an attacker who can compromise 

the system configuration can accomplish nearly any other goal. For example, an 

ordinary extension could be re-assigned into a pool of phones that supervisors can 

listen in on or record conversations for quality control purposes. Damaging or 

deleting information about the IP network used by a VOIP switch results in an 

immediate denial of service. 

The security system itself provides the capabilities for system abuse and misuse. 

That is, compromise of the security system not only allows system abuse but also 

allows the elimination of all traceability and the insertion of trapdoors for 

intruders to use on their next visit. For this reason, the security system must be 

carefully protected. 

83 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

Integrity threats include any in which system functions or data may be corrupted, 

either accidentally or as a result of malicious actions. Misuse may involve 

legitimate users (i.e. insiders performing unauthorized operations) or intruders. 

A legitimate user may perform an incorrect, or unauthorized, operations function 

(e.g., by mistake or out of malice) and may cause deleterious modification, 

destruction, deletion, or disclosure of switch software and data. This threat may 

be caused by several factors including the possibility that the level of access 

permission granted to the user is higher than what the user needs to remain 

functional. 

Intrusion - An intruder may masquerade as a legitimate user and access an

operations port of the switch. There are a number of serious intrusion threats. For 

example, the intruder may use the permission level of the legitimate user and 

perform damaging operations functions such as: 

• disclosing confidential data 

• causing service deterioration by modifying the switch software 

• crashing the switch 

• removing all traces of the intrusion (e.g., modifying the security log) so that it 

may not be readily detected 

Insecure state - At certain times the switch may be vulnerable due to the fact that 

it is not in a secure state. For example: 

• After a system restart, the old security features may have been reset to 

insecure settings, and new features may not yet be activated. (For example, all 

old passwords may have reverted to the default system-password, even though 

new passwords are not yet assigned.) The same may happen at the time of a 

disaster recovery. 

• At the time of installation the switch may be vulnerable until the default 

security features have been replaced.

DHCP Server Insertion Attack 

It is often possible to change the configuration of a target phone by exploiting 

the DHCP response race when the IP phone boots. As soon as the IP phone 

requests a DHCP response, a rogue DHCP server can initiate a response with 

data fields containing false information. 

This attack allows for possible man in the middle attacks on the IP-media 

gateway, and IP Phones. Many methods exist with the potential to reboot the 

84 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

phone remotely, e.g. “social engineering”, ping flood, MAC spoofing (probably 

SNMP hooks, etc.). 

If possible, use static IP addresses for the IP Phones. This will remove the 

necessity of using a DHCP server. Further, using a state based intrusion 

detection system can filter out DHCP server packets from IP Phone ports, 

allowing this traffic only from the legitimate server. 

TFTP Server Insertion Attack 

It is possible to change the configuration of a target phone by exploiting the 

TFTP response race when the IP phone is resetting. A rogue TFTP server can 

supply spurious information before the legitimate server is able to respond to a 

request. This attack allows an attacker to change the configuration of an IP 

Phone. Using a state based intrusion detection system can filter out DHCP 

server packets from IP Phone ports, allowing such traffic only from the 

legitimate server. Organizations looking to deploy VOIP systems should look 

for IP Phone instruments that can download signed binary files. 

A.3 Availability and Denialof Service

Availability refers to the notion that information and services be available for use 

when needed. Availability is the most obvious risk for a switch. Attacks 

exploiting vulnerabilities in the switch software or protocols may lead to 

deterioration or even denial of service or functionality of the switch. For example: 

if unauthorized access can be established to any branch of the communication 

channel (such as a CCS link or a TCP/IP link), it may be possible to flood the link 

with bogus messages causing severe deterioration (possibly denial) of service. A 

voice over IP system may have additional vulnerabilities with Internet 

connections. Because intrusion detection systems fail to intercept a significant 

percentage of Internet based attacks, attackers may be able to bring down VOIP 

systems by exploiting weaknesses in Internet protocols and services. 

Any network may be vulnerable to denial of service attacks, simply by 

overloading the capacity of the system. With VOIP the problem may be 

especially severe, because of its sensitivity to packet loss or delay. 

CPU Resource Consumption Attack without any account information. 

An attacker with remote terminal access to the server may be able to force a system

restart (shutdown all/restart all) by providing the maximum number of characters for 

the login and password buffers multiple times in succession. Additionally, IP Phones

may reboot as a result of this attack. 

In addition to producing a system outage, the restart may not restore uncommitted 

changes or, in some cases, may restore default passwords, which would introduce 

intrusion vulnerabilities. The deployment of a firewall disallowing connections from

85 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

unnecessary or unknown network entities is the first step to overcoming this 

problem. However, there is still the opportunity for an attacker to spoof his MAC 

and IP address, circumventing the firewall protection. 

Default Password Vulnerability 

It is common for switches to have a default login/password set, e.g., 

admin/admin, or root /root. Similarly, VOIP telephones often have default 

keypad sequences that can be used to unlock and modify network information 

This vulnerability would allow an attacker to control the topology of the network 

remotely, allowing for not only complete denial of service to the network, but 

also a port mirroring attack to the attacker’s location, giving the ability to 

intercept any other conversations taking place over the same switch. Further, the 

switch may have a web server interface, providing an attacker with the ability to 

disrupt the network without advance knowledge of switch operations and 

commands. In most systems, telephones download their configuration data on 

startup using TFTP or similar protocols. The configuration specifies the IP 

addresses for Call Manager nodes, so an attacker could substitute another IP 

address pointing to a call manager that would allow eavesdropping or traffic 

analysis. Changing the default password is crucial. Moreover, the graphical 

user interface should be disabled to prevent the interception of plaintext 

administration sessions. 

Exploitable software flaws 

Like other types of software, VOIP systems have been found to have vulnerabilities 

due to buffer overflows and improper packet header handling. These flaws typically 

occur because the software is not validating critical information properly. For 

example, a short integer may be used as a table index without checking whether the 

parameter passed to the function exceeds 32,767, resulting in invalid memory 

accesses or crashing of the system. 

Exploitable software flaws typically result in two types of vulnerabilities: denial of 

service or revelation of critical system parameters. Denial of service can often be 

implemented remotely, by passing packets with specially constructed headers that

cause the software to fail. In some cases the system can be crashed, producing a 

memory dump in which an intruder can find IP addresses of critical system nodes, 

passwords, or other security-relevant information. In addition, buffer overflows that 

allow the introduction of malicious code have been found in VOIP software, as in 

other applications. 

These problems require action from the software vendor, and distribution of patches 

to administrators. Intruders monitor announcements of vulnerabilities, knowing that 

many organizations require days or weeks to update their software. Regular checking 

for software updates and patches is essential to reducing these vulnerabilities. 

86 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

Automated patch handling can assist in reducing the window of opportunity for 

intruders to exploit a known software vulnerability. 

Account Lockout Vulnerability 

An attacker will be able to provide several incorrect login attempts at the telnet

prompt until the account becomes locked out. (This problem is common to most 

password-protected systems, because it prevents attackers from repeating login 

attempts until the correct password is found by trying all possible combinations.) 

The account is unable to connect to the machine for the set lockout time. If 

remote access is not available, this problem can be solved with physical access 

control. 

87 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

B Appendix: VOIP Frequently Asked Questions 

1. What is VOIP? 

Voice Over Internet Protocol is a set of software, hardware, and standards 

designed to make it possible to transmit voice over packet switched networks, 

either an internal Local Area Network, or across the Internet.

2. What are some of the advantages of VOIP? 

a. Cost – a VOIP system is usually cheaper to operate than an equivalent 

office telephone system with a Private Branch Exchange and conventional 

telephone service. 

b. Integration with other services – Innovative services are emerging that 

allow customers to combine web access with telephone features through a 

single PC or terminal. For example, a sales representative could discuss 

products with a customer using the company’s web site. In addition, the 

VOIP system may be integrated with video across the Internet, providing a 

teleconferencing facility.

3. What are some of the disadvantages of VOIP?

a. Startup cost – although VOIP can be expected to save money in the long 

run, the initial installation can be complex and expensive. In addition, a 

single standard has not yet emerged for many aspects of VOIP, so an 

organization must plan to support more than one standard, or expect to 

make relatively frequent changes as the VOIP field develops. 

b. Security – the flexibility of VOIP comes at a price: added complexity in 

securing voice and data. Because VOIP systems are connected to the data 

network, and share many of the same hardware and software components, 

there are more ways for intruders to attack a VOIP system than a 

conventional voice telephone system or PBX. 

4. Can small organizations or home users use VOIP systems?

Yes. Vendors have made VOIP solutions attractive to organizations of all 

sizes, and are now expanding into the home market. Most of the security 

problems discussed in this publication have also extended into the home 

market. 

88 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

5. Can I use my existing network equipment (routers, hubs, etc.) for a VOIP 

network?

No, except possibly for home use. VOIP demands high performance because 

voice communications must be in real time. A few seconds delay in data 

transmission is accepted and common, but a similar delay in a telephone 

conversation would make the system unacceptable to users. VOIP network 

equipment uses special protocols to over come the performance problems 

involved in transmitting voice over the Internet. Existing local area network 

cabling can be used. 

6. Does VOIP require additional phone lines or new phone numbers. 

No, VOIP systems can be tied in to existing connections, 

7. Can I use VOIP like a normal telephone?

Yes, if the right hardware and software is installed. Some basic VOIP 

services are available from messaging programs such as AIM and Microsoft 

Netmeeting, but these programs cannot be used to dial telephone numbers. 

VOIP systems, coupled with a VOIP service provider, enable users to enter 

ordinary telephone numbers to make calls across the Internet. 

8. Do both parties to a call need to have VOIP?

No, VOIP service providers translate VOIP calls to and from a form that is 

carried on the conventional telephone network using IP/TDM gateways, so 

either party may use VOIP or conventional equipment. 

9. How does the sound quality of VOIP compare with traditional systems? 

More recent VOIP systems have sound quality equivalent to conventional 

phones, especially if standalone VOIP phones are used. A “softphone” may 

have packet delay or other quality of service issues if used on a heavily loaded 

PC. 

10. What is a VOIP “softphone”?

The term softphone refers to a telephone capability implemented on an 

ordinary PC, using only special software and a microphone/headset that plugs 

into the PC’s audio ports. As noted in the body of this publication, though, 

softphones should not be used where security or privacy are a concern because 

of the ease with which they can be attacked. These systems are also more 

vulnerable to denial of service attacks from worms and viruses. 

89 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

11. What size PC is required to operate a softphone? 

A high-end PC is not needed. Most systems require only 64MB of RAM and 

a 200 MHz or faster processor. VOIP software is available for all popular 

operating systems. 

12. Will a VOIP system continue to function during a power failure or cable outage? 

If all components have an uninterruptible power supply, the system should 

continue to function as long as the UPS batteries last. However, if the VOIP 

system is implemented on a cable modem, phone service will not be available 

during a cable outage. Similarly, if DSL is used, an outage of the DSL line 

will interrupt phone service. A conventional phone connection or mobile 

phones can serve as a backup. 

90 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

C Appendix: VOIP Terms 

Application Level Gateway (ALG) – Application Level Gateways (ALGs) are 

application specific translation agents that allow an application (like VOIP) on a 

host in one address realm to connect to its counterpart running on a host in 

different realm transparently. An ALG may interact with NAT to set up state, use 

NAT state information, modify application specific payload and perform 

whatever else is necessary to get the application running across disparate address 

realms. 

Abstract syntax notation one (ASN.1): A standard, flexible method that (a) 

describes data structures for representing, encoding, transmitting, and decoding 

data, (b) provides a set of formal rules for describing the structure of objects 

independent of machine-specific encoding techniques, (c) is a formal networkmanagement Transmission Control Protocol/Internet Protocol (TCP/IP) language 

that uses human-readable notation and a compact, encoded representation of the 

same information used in communications protocols, and (d) is a precise, formal 

notation that removes ambiguities. 

 

Call Processor – component that sets up and monitors the state of calls, and 

provides phone number translation, user authorization, and coordination with 

media gateways. 

Codec – coder/decoder, which converts analog voice into digital data and back 

again, and may also compress and decompress the data for more efficient 

transmission. 

Firewall Control Proxy - component that controls a firewall’s handling of a call. 

The firewall control proxy can instruct the firewall to open specific ports that are 

needed by a call, and direct the firewall to close these ports at call termination. 

H.323 - The International Telecommunications Union (ITU) standard for packetswitched network voice and video calling and signaling. 

Jitter - non-uniform delays that can cause packets to arrive and be processed out 

of sequence 

Latency – time delay in processing voice packets. 

Media gateway - the interface between circuit switched networks and IP network. 

Media gateways handle analog/digital conversion, call origination and reception, 

and quality improvement functions such as compression or echo cancellation. 

91 
NIST SP 800-58 Voice Over IP Security 

_____________________________________________________________________________

Media Gateway Control Protocol – common protocol used with media 

gateways to provide network management and control functions. 

PSTN – the public switched telephone network. 

QoS - Quality of Service - a network property that specifies a guaranteed 

throughput level. 

Session Initiation Protocol (SIP) – A standard for voice and video over a packet 

switched network developed by IETF. 

Sniffer – a network monitoring tool, usually a software tool running on a PC. 

92 
NIST SP 800-58 Voice Over IP Security 

_________________________________________________________________________ 

INDEX

advanced encryption standard (AES), 72 

application level gateway (ALG), 59, 77 

authentication, 58, 63, 68, 74, 76, 82 

authentication header (AH), 63, 68, 73 

Availability, 85 

call manager, 3, 14, 53, 86 

confidential, 81 

Confidentiality, 81 

cost, iv, 3, 16, 60, 76, 88 

datagram, 64 

DMZ, 4, 52 

encapsulating security payload (ESP), 

63, 65, 68, 73 

encryption, 3, 17, 18, 19, 20, 24, 63, 64, 

65, 66, 67, 68, 69, 72, 73 

Federal Information Security 

Management Act (FISMA), 10 

file transfer protocol (FTP), 52 

firewall, 4, 20, 24, 36, 37, 52, 53, 56, 57, 

58, 59, 60, 65, 73, 75, 83, 85 

firewall control proxy (FCP), 59, 60 

gateway, 26, 37, 52, 58, 64, 76 

H.245, 26, 27, 28, 37 

H.255, 26 

H.323, 3, 5, 26, 27, 28, 36, 37, 39, 40, 

57, 58, 59, 75, 76, 77, 79 

HTTP, 52, 75, 82 

HTTPS, 83 

IKE, 63, 68, 73 

Integrity, 83 

Internet Engineering Task Force (IETF), 

39, 60, 73 

IP, 3, 4, 13, 15, 21, 23, 37, 45, 46, 52, 

53, 54, 55, 56, 58, 59, 63, 65, 68, 69, 

72, 73, 75, 78, 79, 80, 81, 82, 83, 84, 

85, 86 

IP phone, 53, 69, 83, 84, 85 

IPsec, 4, 21, 24, 55, 62, 63, 64, 65, 66, 

68, 69, 72, 73, 75, 78, 80 

IPsec tunneling, 62, 65 

IPv4, 73 

IPv6, 56, 63, 65, 73, 80 

ISP, 18, 54, 56, 80 

ITU, 19, 26 

jitter, 3, 19, 20, 21, 23, 57, 59, 65, 68, 72 

latency, 3, 19, 20, 21, 22, 23, 24, 37, 57, 

58, 59, 63, 65, 66, 67, 68, 72 

local area network (LAN), 4, 23, 52, 54, 

65, 69, 73 

MAC, 4, 81, 83, 85, 86 

middlebox, 37 

NetMeeting, 13 

network address translation (NAT), 4, 

24, 28, 37, 40, 45, 46, 52, 54, 55, 56, 

57, 58, 59, 60, 62, 68, 73, 75, 79, 80 

Office of Management and Budget 

(OMB) Circular A-130, 10 

packet loss, 3, 19, 21, 22, 23, 65, 67, 68, 

73, 85 

packet sniffer, 63, 65, 82 

proxy, 37, 39, 45, 46, 53 

proxy server, 37, 39, 45, 46, 53 

public switched telephone network 

(PSTN), 13, 16, 18, 19, 22, 26 

quality of service (QoS), 3, 19, 20, 21, 

22, 23, 24, 28, 56, 57, 62, 63, 65, 66, 

67, 68, 72, 73, 76, 78, 89 

Real Time Protocol (RTP), 16, 20, 21, 

23, 28, 40, 44, 45, 53, 56, 57, 58, 59, 

76 

registrar, 39, 46 

router, 54, 56, 58, 64, 67, 69, 83 

session initiation protocol (SIP), 3, 5, 26, 

39, 40, 44, 45, 46, 57, 58, 59, 75, 76, 

77, 79 

stateful firewall, 37, 53, 58 

TCP, 21, 22, 27, 37, 39, 45, 54, 65, 73, 

85 

UDP, 16, 20, 21, 36, 39, 45, 53, 54, 57, 

68, 73 

virtual private network(VPN), 57, 62, 

63, 65, 75, 80 

vulnerabilities, 10, 56, 81, 85, 86 

wide area network (WAN), 4, 23, 52, 65 

Windows, 13, 90 

93