A Survey of VoIP Security Practices in Higher

Education

H. Morrow LongDirector, Information Security

Yale University

NERCOMP Session

Wednesday, March 21, 2007 8:30 a.m. - 9:20 a.m.

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Introductions

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Overview

This presentation will discuss a survey and informal poll of the current campus network VoIP security practices and products in higher education on both wired and wireless networks.

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Agenda

IntroductionWhat is VoIP?VoIP ThreatsVoIP Security ChecklistsVoIP Effective Practices in Higher EdDiscussion and Questions

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VoIP Security Goes Mainstream

In 2006, VoIP Security entered the SANS Top 20 for the first time:

http://www.sans.org/top20/#n1

N1 VoIP Servers and Phones

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VoIP Security Flaws Go Mainstream

2006 VoIP Security vulnerabilities:AsteriskCVE-2006-2898, CVE-2006-4345, CVE-2006-4346, CVE-2006-5444

Cisco Call ManagerCVE-2006-0368, CVE-2006-3594

VoIP PhonesCVE-2005-3717, CVE-2005-3722, CVE-2005-3723, CVE-2006-0305, CVE-2006-0374, CVE-2006-0834, CVE-2006-5038

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VoIP Security Flaws Go Mainstream

2007 VoIP Security vulnerabilities:

AsteriskCVE-2007-1306

Cisco Call Manager / IOS / PIXOSCVE-2007-0648, SA24180/cisco-sa-20070214-

fwsm,SA24179/cisco-sa-20070214-pix

VoIP PhonesCVE-2007-1072, CVE-2007-1062, CVE-2007-

1063

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What is VoIP?

Voice over IP

IP Telephony

Converged Data/Voice Networking

Unified Messaging

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What is VoIP?

2 Major Protocols: H.323 SIP / SIPS

Popular Internet VoIP Proprietary

• Skype• Vonage

Other Zfone/ZRTP (Phil Zimmerman)

Internet Standards related to VoIP Security: IPSEC SSL/TLS SRTP (RFC3711)

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H.323 and SIP

The 2 Major (Local and Enterprise) VoIP Protocols: H.323 SIP

Both protocols:Are hard (but not impossible) to firewallWere not designed for security…Use separate signaling and media (content) channelsUse dynamic portsWere not designed to be NAT “friendly” (embed IP addresses inside signaling/control information)

But: H.323 is more like ISO X. protocols (uses ASN.1/PER) and SIP is more like Internet FTP/SMTP/HTTP/NNTP.

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H.323

Older protocol than SIP, implemented earlierITU Umbrella Standard - built of other H stdsFirst VoIP std to use RTPInteroperates with ISDN PBX systemsUsed by several voice and videoconferencing systemsBuilt into NetMeeting, other commercial and open source programs availableGNU Gatekeeper - accounting/authorization/NAT traversal/H.323 proxy/H.235 security

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H.235 Security

H.235 provides security for H.323Optionally nine security profiles can be used to apply one or more of six security services (authentication, nonrepudiation, integrity, confidentiality, access control, key management) to H.225, H.245 and RTP traffic.

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“Skinny” - Cisco H.323

“Skinny” is Cisco’s lightweight proprietary version of H.3323.SCCP is the acronym for Skinny Client Control Protocol.It is a lower overhead control protocol between the client and Call Manager.

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SIP - Session Initiation Protocol

Overtaking H.323 on LANS - many clients.Created 1996. SIP 2.0 defined in RFC 2543 (1999)-- refined in RFC 3261 (2002).Lightweight, text-based protocol run on top of UDP or TCP (e.g. port 5060- mod P2P model.Uses HTTP “style” status codes & email addresses.Interoperates with XMPP IM (Japper)STUN & newer TURN enable SIP through NAT using public Internet servers.Uses other protocols: SDR, RTP, MGCP, RTSP.Can be stateful/less, client/server or P2P.

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SIP/RTP Architecture

VoIP User-Agent

RTP SIP

SRTP TLS

UDP and TCP

IP

Data Link Layer

Credit: Practical VoIP Security, Syngress

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SIPS - Secure SIP

Secure SIP is a security mechanism defined by SIP RFC 3261 (2002) defines Secure SIP -- a security mechanism using TLS to send SIP messages over an (Transport Layer Security) encrypted channel.Fairly new, competes with IPSEC, VPNs, SRTP -- often referred to as SIP with TLS -- used when IPSEC is overkill or SIP proxies must be used.

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SRTP

Adds message encryption, authentication, integrity and replay protection to to RTP Sister to SRTCP (Secure RTP Control Proto)SRTP/SRTCP encryption, authentication and integrity are independent and can be disabled (“Null” encryption).Single Cipher (AES), 2 modes (counter & feedback modes)External Key mgt (ZRTP, Mikey, …)

Credit: http://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocol

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SRTP Interoperability

Hard IP Phones Avaya, Cisco, Ericsson (&TLS), Siemens, Linksys,

Snom (&TLS)

Soft IP phones Gizmo, Kphone, Snom360 (&TLS), minisip (&TLS)

Hard IP PBX - Alcatel and EricssonSoft-IP-PBX - Asterisk (SIP & H323) and pbxnsipSBC (Session Border Ctrlr) / SIP Firewall Covergence (& SIP & H323) InGate (SIP aware firewall)

Credit: http://en.wikipedia.org/wiki/Secure_Real-time_Transport_Protocol

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Zfone/ZRTP

Created/driven by (Phil Zimmerman)2nd attempt (PGPfhone)Designed to work with current SIP phone programs (via plug-ins).Zfone is the program.ZRTP is an extension to RTP (Real-time Transport Protocol) providing secure real-time transport to secure sessions (SIP, H.323, etc.) already established. Keys are transmitted and managed outside the std signaling. Protection against MitM (man in the Middle) attacks.

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Skype

Peer to Peer ModelSupernodes route traffic for other callsCan be blocked and bandwidth managedOutlawed at some institutionsProprietary strong encryptionNon-CALEA compliance?

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More VoIP Terminology

“Presence” (R U there?)Convergence (Data + Voice = Synergy)Voice MessagingUnified Messaging Systems

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More VoIP Acronyms

ACD Automatic Call Distribution(Call Ctr)IVR Interactive Voice ResponseICE Interactive Connectivity EstablishmentRSVP Resource Reservation Protocol RTSPReal Time Streaming Protocol SDP Session Discovery ProtocolSTUN Simple Traversal of UDP through NATTLS Transport Layer Security (ala SSLv3)TURN Traversal Using Relay NATTTS Text-to-speech server

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Non-Cyber Security-related VoIP Issues

911 - where does 911 ring?E-911 - need to provide location information?Emergency access - during network or power outages

• Use Power-over-Ethernet (PoE AKA IEEE 802.3af) cabling

• Provide at least the minimal # of land lines per # rooms (e.g. or as required by law)

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PBX System Components

PSTNEndpoints (Phones, Faxes, Modems.)Lines (e.g. Station lines)TrunksRemote PBXesAdjuncts (VM, ACD, IVR, …)CDR (Call Detail Recording)Voice/PBX Firewalls

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VoIP System Components

Media Gateways -- e.g. to PSTN/PBXesEndpoints (User Agents): softphones, IM/Video/VoIP/ATA (Analog Telephone Adatper) Media Servers (VM, ACD, IVR, TTS,VC)H.323 GatekeepersSIP Registration, Redirect ServersSIP Proxy Servers Firewalls/ALGs

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VoIP Threats

VoIP Networks have many of the same threats to security, privacy and reliability as data networks do, but they also bring in the problems of the telephone system and have some special threats all their own.

Converged networks can combine threats from the data and VoIP world -- making the new network less secure (in the opinion of some).

Data network people are afraid VoIP infrastrucutre will weaken the security of their data network and the voice/telecom people feel the same about data / IP networks.

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Other VoIP Architectures

SkypeIAXH.248Microsoft Live Communication Server 2005 (MLCS) TLS between client and server Mutual TLS server-to-server

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VoIP vs. PSTN

Remember that “POTS” telephones have little security -- ordinary phone conversations are not encrypted and can be tapped or eavesdropped.You can actually have better security using VoIP IF you use strong encryption (and a good implementation).

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VoIP Threats

DDoS / DoS Attacks ICMP Flood (eg ‘pings’) to Phone or Call Mgr

Unauthorized AccessToll FraudVoicemail hackingEavesdropping (Call and/or Control)Call HijackingApplication Level Attacks

Credit: Juniper Networks

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IP Network Threats

Ethernet, IP and DNS address spoofingARP and DNS Cache PoisoningQuantity-based packet floodingStack DoS attacksVLAN “jumping”QoS / prioritization attacks

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Organizing VoIP Threats

Standard IP Network Threats(to the CIA triad)

C - ConfidentialityI - IntegrityA - Availability

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Organizing VoIP Threats

Advanced IP Network Issues/challenges

(triple A)A - AuthenticationA - AuthorizationA - Accounting

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Application-Specific VoIP Threats

“Phone” spoofing - registering a SIP client with someone else's identifier (no auth.). a successful attack would cause the similarly

registered phone to ring when someone called the legitimate owner of the number.

Credit: Jeremy George, Yale University

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Threat to Confidentiality

Programs exist to listen to SIP and other VoIP streams (and record them).It is possible to capture packets on switched networks (by overflowing ARP tables, poisoning ARP caches, etc.).Encryption should be used but has side- effects: : on latency, on sound qulaity (packetization and compression chunking can lead to clipped staccato speech).

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Application-Specific VoIP Threats

Caller-ID / ANI “Spoofing” (faking source #) Trivial to do -- don’t trust Caller-ID -- OK to

screen w/

Credit: Jeremy George, Yale University

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Threat to Integrity

It is possible to ‘hijack’ sessions.It is possible to modify voice over IP streams.Once again, use encryption (or at least cryptographic integrity checks) to prevent this.

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Application-Specific VoIP Threats

MitM “spoofing” CALEA is a ‘legit’ application of this. DoS attacks are known immediately by

communicating parties Call content is neither overheard nor compromised. Some proxies have logic in them that identifies a

likely DoS attack and discard those packets (ask your vendor!).

Encryption is the best proection against MitM spoofing.

Credit: Jeremy George, Yale University

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Threats to Availability

Quality of Service (QoS) problems: Latency - time for traffic to go from source

to destination (one-way and round-trip).150ms is Max RTT for PSTN. VoIP at 400ms is at outer limit of tolerable range.

Jitter - variability in latency and out-of-order packet arrival times. Buffering can help here.

Packet Loss - results in gaps in communication.

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Application-Specific VoIP Threats

“Special DoS (Denial of Service) attacks high volume flood of SIP INVITEs high volume flood of SIP REGISTER commands Control Packet / Call Data Floods Packet Replay / Injection / Modification

Credit: Jeremy George, Yale University

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Application-Specific VoIP Threats

“BID attacks on SIPS” Get SIPS devices to downgrade to

ordinary SIP

Credit: Jeremy George, Yale University

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Application-Specific VoIP Threats

Rogue SIP Proxies Impersonate a proxy to a User-Agent

Credit: Practical VoIP Security, Syngress

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VoIP Security Checklist

Practical VoIP Security “high level short list”:Create, publish and enforce security policies.Practice rigorous physical security.Verify user identities.Actively monitor logs, firewalls & IDSes.Logically segregate data & voice traffic.Harden Oses.Encrypt whenever and whatever you can.

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VoIP Security Checklist

Juniper Best Pracices Security Measures1. Maintain Current Patch Levels 2. Install a Good Anti-Virus System and Update it Regularly3. Apply State-of-the-Art Intrusion Detection and Prevention Systems4. Install Application-Layer Gateways between Trusted and Untrusted

Zones. 5. Enforce SIP security by means of Authentication, Authorization and IPSec6. Establish Policy-Based Security Zones to Isolate VoIP Segments. 7. Run VoIP Traffic on VPNs to Minimize Eavesdropping Risk on Critical

Segments. 8. Use VLANs to Prioritize and Protect Voice Traffic from Data Network

Attacks9. Apply Encryption Selectively10 Protect Against UDP Flooding11. Develop a Holistic Security Program

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Metagroup Checklist

IP Telephony-Specific Security FeaturesThe Call Control Server

Harden/Strip down OS.Use secure OS.Authenticate & authorize all user & device access to servers.Require strong authentication for all configuration and software upgrades.Should support app level signaling message auth.Should support call setup info encryption.

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Metagroup Checklist

IP Telephony-Specific Security FeaturesThe Voice Gateway:

Require strong authentication for all configuration and software upgrades .Provide DoS protection on IP inteface.Should be configured to route calls only via the call control server.Secure OS w/anti-virus AND host-based IDS.Should support call setup info and media (voice content) encryption.Should support a media (voice content) protocol authentication on a per-packet basis.

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Metagroup Checklist

IP Telephony-Specific Security FeaturesThe IP Phone:

Must authenticate itself to the call control server or a proxy server upon initial registrationMust support strong authentication for any remote configuration and software upgrades .Should support a configurable access control list to control any incoming traffic (e.g. H.323/SIP, RTP, HTTP, FTP, DHCP)..When supporting an additional Ethernet port for PC connectivity, should have this implemented via a switching function combined with VLAN functionality.Should support encryption of both call setup info and media as needed. Using encryption can add an additional end-to-end delay on each media packet.

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VoIP Security Checklist

Detailed and Specfic list:Use a separate VLAN with 802.1p/q QoS w/priority VLAN tagging for the VoIP network.Use a private (RFC1918) IP network for the VoIP LAN.Use NAT and/or proxies to hide internal addresses.Use a firewall (packet filtering or ALG) to protect & connect the VoIP network to the data IP network.Use an IDS or IPS to examine the traffic allowed through the firewall (may be built into the firewall).Use TLS to protect SIP and SRTP to protect RTP.Use NAC, 802.1X & RADIUS auth & SIP-aware FW.

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Listservs & Newsgroups

EDUCAUSE Security Discussion Listservhttp://www.educause.edu/SecurityDiscussionGroup/979

VOIPSA Best Practices Working Grouphttp://www.voipsa.org/Activities/bestpractices.phpVOIPSA Best Practices WG List http://voipsa.org/mailman/listinfo/bestpractices_voipsa.org

NIST Publication Mailing listhttp://csrc.nist.gov/compubs-mail.html

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VoIP Security Effective Practices in Higher Ed

One anonymous school:Uses separate VLAN, L2 switches and RFC1918 IP addresses for VoIP network.Provides separate connections (and bandwidth) to each building with VoIP.Softphones can participate from regular campus LAN (aren’t required to use a 2nd NIC on the VoIP network).

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VoIP Security Effective Practices in Higher Ed

A 2nd anonymous school:

Has enterprise polycom gateways (a bunch of them) that have priority in QOS on the routers..Allows traffic via ports inbound on the above routers for this ‘legit’ traffic. Doesn’t restrict H.323.Blocks SIP and Vonage because they don’t open the inbound ports.Packet8 and other SIP applications which use STUN work fine (because of tunneling).Skype is a problem (paritcularly Supernodes at times).

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Wrap-Up

Question & Answer

Session Evaluation & Feedback

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Contact Info

H. Morrow Long

morrow.long@yale.edu

Security.yale.edu

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Credits:

Cisco - Configuring SIP High Availability Applications, http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/callc_c/sip_c/sipha_c/hachap2.htm

Jeremy George, Yale University, ““SIP.edu Cookbook - Security Considerations”http://mit.edu/sip/sip.edu/security.shtml

Deb Shinder, 2006/12/1 “Make a SIP-based VoIP network more secure”, http://articles.techrepublic.com.com/5100-1035_11-6145231.html?part=rss&tag=feed&subj=tr

Deb Shinder, 2007/1/7 “Take a multi-layered approach to VoIP security”, http://articles.techrepublic.com.com/5100-1035_11-6145231.html?part=rss&tag=feed&subj=tr

Jose J. Valdes, Jr., Colorado State University “Voice over Internet Protocol (VoIP) Security”, Net@Edu Conference, ICS – Wireless Group Meeting, Tempe, Arizona, February 6, 2005

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Credits:

Practical VoIP Security by Larry Chaffin, Jan Kanclirz, Jr., Thomas Porter, Choon Shim, Andy Zmolek, Syngress, March 2006

Wikipedia (pages on H.323, SIP, SRTP, ZRTP), Zfone, etc.)