IEEE NJ Coast Section Seminar on Wireless LAN & IP Telephony Session

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IEEE NJ Coast Section Seminar on Wireless LAN & IP Telephony Session I5 Creating Secure Services for Internet Telephony Henning Schulzrinne Columbia University [email protected]

Overview What are IP telephony services? Where do services reside? How to create services? – – – – – basic “fixed” services (call forwarding, follow me, .) registration-based services: caller preferences sip-cgi model Call Processing Language (CPL) sip servlets & JAIN Event notification and presence Example of an enterprise IP telephony platform Billing in IP telephony IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 2

Overview Security in IP telephony – dealing with NATs and firewalls – differences to classical PSTN networks – threats theft of service registration impersonation denial of service privacy – current SIP approaches Summary and conclusion IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 3

Aside: evolution of SIP Not quite what we had in mind – initially, SIP for initiating multicast conferencing in progress since 1992 still small niche even the IAB and IESG meet by POTS conference – then VoIP written-off equipment (circuit-switched) vs. new equipment (VoIP) bandwidth is (mostly) not the problem “can’t get new services if other end is POTS’’ “why use VoIP if I can’t get new services” IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 4

Evolution of SIP VoIP: avoiding the installed base issue – cable modems – lifeline service – 3GPP – vaporware? Finally, IM/presence and events – probably, first major application – offers real advantage: interoperable IM – also, new service IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 5

VoIP at Home Lifeline (power) Multiple phones per household – – – – expensive to do over PNA or 802.11 BlueTooth range too short need wireless SIP base station handsets PDAs with 802.11 and GSM? (Treo ) Incentives – SMS & IM services IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 6

SIP phones Hard to build really basic phones – need real multitasking OS – need large set of protocols: IP, DNS, DHCP, maybe IPsec, SNTP and SNMP UDP, TCP, maybe TLS HTTP (configuration), RTP, SIP – user-interface for entering URLs is a pain see “success” of Internet appliances “PCs with handset” cost 500 and still have a Palmsize display thus, offer services – Java-programmable – XML forms input IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 7

Example SIP phones IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 8

What are IP telephony services? Services (features) modify basic call behavior Can be – invoked by user – pre-programmed into network elements (e.g., SIP proxies) – programmable feature logic PSTN: CLASS (Custom local area signaling services) features – – – – – – call waiting call forwarding caller ID (calling number delivery) distinctive ringing selective call rejection three-way calling, . PSTN: pre-subscribed for feature access codes (e.g., *66) IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 9

IP telephony services Call routing services: precall, one party – – – – – – – – – – speed dial click-to-dial call forwarding “follow me” call filtering/blocking (in/out) do not disturb distinctive ringing call prioritization feature-based agent selection call return Call handling features – hotline – autoanswer – intercom Multi-party features – call waiting – whispered call waiting – blind transfer: no confirmation of success – attended transfer – consultative transfer: three-party conference transfer – conference call – call park – call pickup – music on hold – call monitoring – barge-in – speakerphone paging – single-line extension IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 10

IP telephony features – Internet-specific Presence-enabled calls – place call only if callee is available Presence-enabled conferencing – call conference participants when all are online and not busy IM conference alerts – receive IM when someone joins a conference Unified messaging – receive email with new voice message – IM alert for voicemails IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 11

Voice-enabled features Interactive Voice Response (IVR) – VoiceXML – voice browser IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 12

Voice-enabled features: VoiceXML ?xml version "1.0"? vxml version "2.0" form id "basic" field name "acctnum" type "digits" prompt What is your account number? /prompt /field field name "acctphone" type "phone" prompt What is your home telephone number? /prompt filled !-- The values obtained by the two fields are supplied to the calling dialog by the "return" element. -- return namelist "acctnum acctphone"/ /filled /field /form /vxml IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 13

PSTN vs. Internet Telephony Internet Telephony end system PSTN Number of lines or pending calls is virtually unlimited Single line, 12 buttons and hook flash to signal IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony More intelligence, PCs can be considered to be end-user devices March 28, 2002 14

PSTN vs. Internet Telephony PSTN: Signaling & Media Internet telephony: Signaling & Media Signaling Signaling Media IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 15

Service provider architectures Models of providing services: – IP PBX – IP Centrex (and cable/DSL) – Carrier / 3G Similar equipment (logically), but – different trust models – sharing of resources (SIP proxies, gateways) IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 16

IP PBX IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 17

IP Centrex IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 18

IP Carrier IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 19

3G Architecture (Registration) mobility management signaling serving CSCF interrogating interrogating proxy home IM domain registration signaling (SIP) visited IM domain IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 20

Service models & protocols Master-slave protocols (MGCP, Megaco) – feature logic in media gateway controller (MGC) – send detailed behavioral commands to MG send ring tone expect dialed digit string play announcement – MG can only “guess” what is meant – assembly-language instructions Peer-to-peer protocols (SIP, H.323) – more like function calls – methods (SIP method, H.323 request) and parameters (SIP headers, H.323 ASN.1 variables) – H.323: per-feature specification (H.450.x) – SIP: building blocks (Headers, REFER, JOIN, .) IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 21

Combining peer-to-peer and master-slave IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 22

CLASS services: Caller-ID SIP To/From headers ( Organization) Also: Call-Info Call-Info: http://alice.com/photo.jpg ;purpose icon, http://alice.com/ ;purpose info Can be “anonymous’’ Cannot necessarily be trusted, since inserted by user Remote-Party-ID: "John Doe" sip:[email protected] ;party calling; idtype subscriber;privacy full;screen yes IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 23

CLASS services: call forwarding, follow-me Built into core SIP Call forwarding: – either at proxy or at end system – 302 Contact: temporary forwarding – 301 Contact: permanent forwarding Follow me: – REGISTER using single identifier – with different temporary IP addresses – “adopt” different hardware via (e.g.,) i-button IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 24

SIP personal mobility IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 25

Call filtering (in/out) Outbound call filtering done by outbound proxy Often, outbound proxy controls firewall Inbound call filtering at any of the stages: – e.g., sip:[email protected] sip:[email protected] – proxies can do filtering at bigcorp.com eng.bigcorp.com paris.eng.bigcorp.com Fixed or programmable rules (later) IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 26

Call routing -- forking IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 27

Call routing -- ENUM Translation between E.164 telephone numbers and URIs (e.g., SIP URIs) RFC 2916 46-8-9761234 becomes 4.3.2.1.6.7.9.8.6.4.e164.arpa Look up using (new) NAPTR DNS record Example contact 1st using SIP, 2nd using email: ORIGIN 4.3.2.1.6.7.9.8.6.4.e164.arpa. IN NAPTR 100 10 "u" "sip E2U" "! .* ! sip:[email protected]!" . IN NAPTR 102 10 "u" "mailto E2U" "! .* ! mailto:[email protected]!" . IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 28

Call routing – TRIP and SLP TRIP (RFC 3219) allows routing of SIP requests to the “best” IP telephony gateway Based on BGP model of route propagation IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 29

Do not disturb & distinctive ringing End system or proxy features Distinctive ringing inserted by proxy: Alert-Info: http://www.example.com/sounds/moo.wav Do not disturb: – 600 (Busy) – 603 (Decline) – with Retry-After IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 30

Call prioritization SIP Priority header Subject: A tornado is heading our way! Priority: emergency Can be inserted or removed by proxy Useful for call routing IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 31

Caller preferences One SIP address many destinations: – home vs. office – cell phone vs. landline – PC video phone vs. black phone Callee’s proxy decides, but caller preferences mechanism allows caller to influence choices Can influence: – – – – – whether to proxy or redirect which URI to proxy or redirect to whether to fork or not whether to search recursively or not whether to search in parallel or sequentially IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 32

Caller preferences Adds parameters to Contact headers describing properties of location: Carol speaks English, Spanish and German and can send/receive audio video, but only wants this address to be used for urgent calls: Contact: Carol sip:[email protected] ;language "en,es,de" ;media "audio/*,video/*,application/chat" ;duplex "full" ;priority "urgent“ INVITE request then contains headers: Accept-Contact: sip:user@host;feature "voicemail&attendant" Accept-Contact: sip:[email protected];mobility "! fixed" IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 33

Using URIs for SIP Service Control RFC 3087 User part is left to local configuration Voice mail services sip:[email protected];mode deposit sip:[email protected] Ad-hoc conferences Invoke VoiceXML scripts sip:dialog.vxml.http%3a//dialogs.server.com/ [email protected] IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 34

Using SIP events for services Many telecom services generate asynchronous events: – participant joined or left conference – message waiting – call leg completed or terminated SIP defines event notification requests: SUBSCRIBE and NOTIFY Event packages for call legs, conferences, message waiting, IM, DTMF, . NOTIFY sip:[email protected] SIP/2.0 To: sip:[email protected] ;tag 78923 From: sip:[email protected] ;tag 4442 Event: message-summary Content-Type: application/simple-message-summary Messages-Waiting: yes Voicemail: 4/8 (1/2) IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 35

Call waiting no notion of “lines” unlimited number of line presences Line 2 ringing Wait 2 minutes Press line 2 A Talk on line 1 INVITE, SDP’s c 0 180 Ringing INVITE 182 Wait 2 minutes 200 OK B C IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 36

Call waiting A Hold on line 1 200 OK Talk on line 2 C B IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 37

Call transfer (unsupervised) B1 3 BYE A 1 REFER B2 Referred-By: B1 A 2 INVITE B2 Referred-By: B1 B2 IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 38

Multi-party features Permanently or temporarily mixing multiple media streams Generally, combinations of – adding conference servers (ad-hoc conferences) – transfer: use REFER to ask other party to do something – combinations of who asks whom to do what recipient just follows instructions IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 39

Third-party call control Separate signaling and media endpoints Also sometimes called back-to-back UA (B2BUA) but some B2BUA’s handle media, too RTP 200 200 3 INVITE 2 SDP (from 2) 5 6 ACK 4 SDP (from 4) INVITE no SDP 1 SIP ACK SIP IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 40

End system vs. Network server Network server End system Permanent IP address Always on (User can have unique address and can always be reached) Temporary IP address Powered off so often (User’s address always changed and can not be reached sometime) Ample computational capacity High bandwidth (Conference) Limited computational capacity Low bandwidth (One to one or small size conf.) Indirect user interaction Usually only deals with signaling (Based on predefined mechanisms, or indirect user interaction, like through web page) Direct user interaction Signal and media converge (easier to deal with human interaction, easier to deal with interaction with media) IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 41

End system vs. Network server Network server End system Information hiding Busy handling Logical call distribution Call transfer Gateway Distinctive ringing IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 42

Service location examples Service End system Network (proxy) Network with Media (UA) Distinctive ringing Yes Can assist Can assist Visual call id Yes Can assist Can assist Call waiting Yes No Yes(*) CF busy Yes Yes(*) Yes(*) CF no answer Yes Yes Yes CF no device No Yes Yes Location hiding No Yes Yes Transfer Yes No No Conference bridge Yes No Yes Gateway to PSTN No No Yes Firewall control No No Yes Voicemail Yes No Yes IEEE NJ Coast Section seminar on Wireless LAN & IP (*) with information provided by end system Telephony March 28, 2002 43

Service architecture Programming language model Service Logic Programming Interface Requests Requests SIP Server Function Responses IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony Responses March 28, 2002 44

Programmable service creation Can’t win by (just) recreating PSTN services Programmable services: – – – – equipment vendors, operators: JAIN local sysadmin, vertical markets: sip-cgi proxy-based call routing: CPL voice-based control: VoiceXML IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 45

Programmable service creation API servlets sip-cgi CPL languageindependent no Java only yes own secure no mostly can be yes end user service creation no yes power users yes GUI tools no no no yes Multimedia some yes yes yes call creation yes no no no IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 46

APIs (e.g., JAIN) Tradition of TAPI, JTAPI, . Typically, call model Treat calls as objects to be manipulated e.g., JAIN: – bearer independent (PSTN, IP, ATM) – protocol-independent (ISUP, SIP, H.323, BICC, .) – protocol APIs and application APIs IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 47

SIP servlets Servlet runs in SIP server Receives SIP objects and processes them Example: call rejection application import org.ietf.sip.*; public class RejectServlet extends SipServletAdapter { protected int statusCode; protected String reasonPhrase; public void init(ServletConfig config) { super.init(config); try { statusCode Integer.parseInt(getInitParameter("status-code")); reasonPhrase getInitParameter("reason-phrase"); } catch (Exception ) {.} } public boolean doInvite(SipRequest req) { SipResponse res req.createResponse(); res.setStatus(statusCode, reasonPhrase); res.send(); return true; } } IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 48

sip-cgi web common gateway interface (cgi): – oldest (and still most commonly used) interface for dynamic content generation – web server invokes process and passes HTTP request via stdin (POST body) environment variables HTTP headers, URL arguments as POST body or GET headers (?arg1 var1&arg2 var2) – new process for each request not very efficient – but easy to learn, robust (no state) – support from just about any programming language (C, Perl, Tcl, Python, VisualBasic, .) Adapt cgi model to SIP sip-cgi RFC 3050 IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 49

sip-cgi Designed for SIP proxies and end systems: – – – – call routing controlling forking call rejection call modification (Priority, Call-Info, Alert-Info) cgi: once per HTTP request sip-cgi: maintain state via an opaque token script gets body of request on stdin script gets SIP headers via environment variables initiates actions via stdout: – – – – proxy request return response generate request generate response IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 50

sip-cgi examples Block *@vinylsiding.com: if (defined ENV{SIP FROM} && ENV{SIP FROM} "sip:*@vinylsiding.com") { print "SIP/2.0 600 I can't talk right now\n\ n"; } Make calls from boss urgent: if (defined ENV{SIP FROM} && ENV{SIP FROM} /sip:[email protected]/) { foreach reg (get regs()) { print "CGI-PROXY-REQUEST reg SIP/2.0\n"; print "Priority: urgent\n\n"; } } IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 51

Call Processing Language (CPL) XML-based “language” for processing requests intentionally restricted to branching and subroutines no variables, no loops thus, easily represented graphically mostly used for SIP, but protocol-independent integrates notion of calendaring (time ranges) structured tree describing actions performed on call setup event top-level events: incoming and outgoing IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 52

CPL Location set stored as implicit global variable – operations can add, filter and delete entries Switches: – – – – address language time, using CALSCH notation (e.g., exported from Outlook) priority Proxy node proxies request and then branches on response (busy, redirection, noanswer, .) Reject and redirect perform corresponding protocol actions Supports abstract logging and email operation IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 53

CPL example busy Call String-switch field: from location url: sip:jones@ example.com proxy timeout: 10s timeout failure match: *@example.com otherwise location url: sip:jones@ voicemail. example.com merge: clear IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony redirect March 28, 2002 54

CPL example ?xml version "1.0" ? !DOCTYPE call SYSTEM "cpl.dtd" cpl incoming lookup source "http://www.example.com/cgi-bin/locate.cgi? user jones" timeout "8" success proxy / /success failure mail url "mailto:[email protected]&Subject lookup %20failed" / /failure /lookup /incoming /cpl IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 55

CPL example: anonymous call screening cpl incoming address-switch field "origin" subfield "user" address is "anonymous" reject status "reject" reason "I don't accept anonymous calls" / /address /address-switch /incoming /cpl IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 56

Billing PSTN: evolution from distance/time-sensitive perminute billing – bucket of minutes – flat-rate plans (“all you can eat”): Canada, AT&T Per-minute billing doesn’t fit well: – SIP sessions can remain open for months, without sending a single packet – voice silence suppression unfair to charge for both directions for large conferences utility – incremental value is non-linear – thus, video unlikely bit rate IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 57

Billing and charging What are we billing for? – infrastructure – services unlikely to be able to charge for call forwarding for corporate users but Yahoo might for residential users – traffic but network cost depends on peak usage, not average usage treat all traffic the same? 3G: charge more for data traffic than voice traffic? – escalation of traffic cloaking and detection A simple billing model – bill per-minute for calls gatewayed into the PSTN – bill for services on a subscription basis (e.g., as part of ISP service) – bill for traffic independent of traffic type by volume, 95th percentile, congestion pricing IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 58

Open Settlement Protocol (OSP) clearing-house model IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 59

AAA Authentication, Authorization, Accounting separate SIP protocol elements from making authentication/authorization decisions allow visited proxy to ask home proxy of visitor whether visitor is legit accounting: – resource dimensioning – apportionment of charges – commercial billing three primary protocols: – RADIUS – used for dial-up servers, popular with ISPs can lose data (UDP) – DIAMETER – successor of RADIUS will be used in 3G for AAA IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 60

Challenges: Security Classical model of restricted access systems cryptographic security Objectives: – – – – identification for access control & billing phone/IM spam control (black/white lists) call routing privacy IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 61

SIP security Bar is higher than for email – telephone expectations (albeit wrong) SIP carries media encryption keys Potential for nuisance – phone spam at 2 am Safety – prevent emergency calls IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 62

System model outbound proxy SIP trapezoid [email protected] : 128.59.16. 1 registrar IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 63

Threats Bogus requests (e.g., fake From) Modification of content – REGISTER Contact – SDP to redirect media Insertion of requests into existing dialogs: BYE, reINVITE Bid-down attacks: attacker gets to pick algorithm Denial of service (DoS) attacks Privacy: SDP may include media session keys Inside vs. outside threats Trust domains – can proxies be trusted? IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 64

Threats third-party – not on path – can generate requests passive man-in-middle (MIM) – listen, but not modify active man-in-middle replay cut-and-paste IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 65

L3/L4 security options IPsec Provides keying mechanism but IKE is complex and has interop problems works for all transport protocol (TCP, SCTP, UDP, ) no credential-fetching API TLS provides keying mechanism good credential binding mechanism no support for UDP; SCTP in progress IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 66

Hop-by-hop security: TLS Server certificates well-established for web servers Per-user certificates less so – email return-address (class 1) certificate not difficult (Thawte, Verisign) Server can challenge client for certificate last-hop challenge IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 67

HTTP Digest authentication Allows user-to-user (registrar) authentication – mostly client-to-server – but also server-to-client (Authentication-Info) Also, Proxy-Authenticate and Proxy-Authorization – May be stacked for multiple proxies on path IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 68

HTTP Digest authentication 401 Unauthorized WWW-Authenticate: Digest realm "[email protected]", qop auth, nonce "dcd9" REGISTER To: sip:[email protected] REGISTER To: sip:[email protected] Authorization: Digest username "alice", nc 00000001, cnonce "defg", response "9f01" REGISTER To: sip:[email protected] Authorization: Digest username "alice", nc 00000002, cnonce "abcd", response "6629" IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 69

End-to-end authentication What do we need to prove? – – – – Person sending BYE is same as sending INVITE Person calling today is same as yesterday Person is indeed "Alice Wonder, working for Deutsche Bank" Person is somebody with account at MCI Worldcom IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 70

End-to-end authentication Why end-to-end authentication? – prevent phone/IM spam – nuisance callers – trust: is this really somebody from my company asking about the new widget? Problem: generic identities are cheap – filtering [email protected] doesn't prevent calls from [email protected] (new day, sam person) IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 71

End-to-end authentication and confidentiality Shared secrets – only scales (N2) to very small groups OpenPGP chain of trust S/MIME-like encapsulation – CA-signed (Verisign, Thawte) every end point needs to have list of Cas need CRL checking – ssh-style IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 72

Ssh-style authentication Self-signed (or unsigned) certificate Allows active man-in-middle to replace with own certificate – always need secure (against modification) way to convey public key However, safe once established IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 73

DOS attacks CPU complexity: get SIP entity to perform work Memory exhaustion: SIP entity keeps state (TCP SYN flood) Amplification: single message triggers group of message to target – even easier in SIP, since Via not subject to address filtering IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 74

DOS attacks: amplification Normal SIP UDP operation: – one INVITE with fake Via – retransmit 401/407 (to target) 8 times Modified procedure: – only send one 401/407 for each INVITE Suggestion: have null authentication – prevents amplification of other responses – E.g., user "anonymous", password empty IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 75

DOS attacks: memory SIP vulnerable if state kept after INVITE Same solution: challenge with 401 Server does not need to keep challenge nonce, but needs to check nonce freshness IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 76

Challenges: NATs and firewalls NATs and firewalls reduce Internet to web and email service – – – – firewall, NAT: no inbound connections NAT: no externally usable address NAT: many different versions - binding duration lack of permanent address (e.g., DHCP) not a problem - SIP address binding – misperception: NAT security IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 77

Challenges: NAT and firewalls Solutions: – longer term: IPv6 – longer term: MIDCOM for firewall control? control by border proxy? – short term: NAT: STUN and SHIPWORM send packet to external server server returns external address, port use that address for inbound UDP packets IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 78

Emergency calls Opportunity for enhanced services: – video, biometrics, IM Finding the right emergency call center (PSAP) – VoIP admin domain may span multiple 911 calling areas Common emergency address User location – GPS doesn’t work indoors – phones can move easily – IP address does not help IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 79

Emergency calls common emergency identifier: sos@domain EPAD REGISTER sip:sos 302 Moved Contact: sip:[email protected] Contact: tel: 1-201-911-1234 Location: 07605 INVITE sip:sos SIP proxy Location: 07605 IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony INVITE sip:[email protected] Location: 07605 March 28, 2002 80

Scaling and redundancy Single host can handle 10-100 calls registrations/second 18,000-180,000 users – 1 call, 1 registration/hour Conference server: about 50 small conferences or large conference with 100 users For larger system and redundancy, replicate proxy server IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 81

Scaling and redundancy DNS SRV records allow static load balancing and failover – but failed systems increase call setup delay – can also use IP address “stealing” to mask failed systems, as long as load 50% Still need common database – can separate REGISTER – make rest read-only IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 82

Large system stateless proxies sip1.example.co m a1.example.co m a2.example.co m sip2.example.co m sip:[email protected] om sip:[email protected] om sip3.example.co m sip. udp SRV 0 0 sip1.example.com b1.example.co m b2.example.co m sip. udp SRV 0 0 b1.example.com 0 0 sip2.example.com 0 0 b2.example.com 0 0 sip3.example.com IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 83

Enterprise VoIP Allow migration of enterprises to IP multimedia communication Add capacity to existing PBX, without upgrade Allow both – IP centrex: hosted by carrier – “PBX”-style: locally hosted – Unlike classical centrex, transition can be done transparently IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 84

Motivation Not cheaper phone calls Single number, follow-me – even for analog phone users Integration of presence – person already busy – better than callback – physical environment (IR sensors) Integration of IM – no need to look up IM address – missed calls become IMs – move immediately to voice if IM too tedious IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 85

Migration strategy Add IP phones to existing PBX or Centrex system – PBX as gateway 1. – 2. 3. 4. Initial investment: 2k for gateway Add multimedia capabilities: PCs, dedicated video servers “Reverse” PBX: replace PSTN connection with SIP/IP connection to carrier Retire PSTN phones IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 86

Example: Columbia Dept. of CS About 100 analog phones on small PBX – DID – no voicemail T1 to local carrier Added small gateway and T1 trunk Call to 7134 becomes sip:7134@cs Ethernet phones, soft phones and conference room CINEMA set of servers, running on 1U rackmount server IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 87

CINEMA components Cisco 7960 MySQL user database LDAP server sipconf rtspd conferencing server (MCU) RTSP media server sipd proxy/redirect server RTSP plug'n'sip wireless 802.11b unified messaging server Pingtel Nortel Meridian PBX T1 sipum Cisco 2600 T1 VoiceXML server SIP sipvxml PhoneJack interface sipc SIP-H.323 converter sip-h323 IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 88

Experiences Need flexible name mapping – Alice.Cueba@cs alice@cs – sources: database, LDAP, sendmail aliases, Automatic import of user accounts: – In university, thousands each September /etc/passwd LDAP, ActiveDirectory, – much easier than most closed PBXs Integrate with Ethernet phone configuration – often, bunch of tftp files Integrate with RADIUS accounting IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 89

Experiences Password integration difficult – Digest needs plain-text, not hashed Different user classes: students, faculty, admin, guests, Who pays if call is forwarded/proxied? – authentication and billing behavior of PBX and SIP system may differ – but much better real-time rating IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 90

SIP doesn’t have to be in a phone IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 91

Event notification Missing new service in the Internet Existing services: – get & put data, remote procedure call: HTTP/SOAP (ftp) – asynchronous delivery with delayed pick-up: SMTP ( POP, IMAP) Do not address asynchronous (triggered) immediate IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 92

Event notification Very common: – – – – operating systems (interrupts, signals, event loop) SNMP trap some research prototypes (e.g., Siena) attempted, but ugly: periodic web-page reload reverse HTTP IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 93

SIP event notification Uses beyond SIP and IM/presence: – Alarms (“fire on Elm Street”) – Web page has changed cooperative web browsing state update without Java applets – Network management – Distributed games IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 94

Conclusion Service creation as central reason for IP telephony Beyond replication of PSTN services: – – – – – modularity easy interface to external databases user-created services interface to web services (SOAP) event model as versatile service component Security as core component – protect users against impersonation, phone/IM spam – user privacy – operator protection often secondary unless SIP is used in billing Deploying SIP services – example of a PBX-like service IEEE NJ Coast Section seminar on Wireless LAN & IP Telephony March 28, 2002 95

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