What is it? SIPSAK: –It’s a small command line tool for developers and administrators of Session Initiation Protocol applications. Try it on FreeBSD: –/usr/ports/net/sipsak Web site: –sipsak.org
How to use it? man sipsak Send an OPTIONS request to firstname.lastname@example.org and display received replies –sipsak -vv -s sip:email@example.com Send the instant message "Lunch time!" to the colleague and show result: –sipsak -M -v -s sip:colleaue@work -B "Lunch time!"
What problems do users face? 四面楚歌
SIP Security SIP security is a vast and challenging field. Authentication – Can users steal other users identity? Integrity – Is the SIP message received the same as the one sent? Confidentiality – Is someone else listening on your SIP call setup?
Threats Fake requests (e.g., fake From) Modification of content –R–REGISTER Contact –S–SDP to redirect media Insertion of requests into existing dialogs: BYE, re-INVITE Denial of service (DoS) attacks Privacy Trust domains – can proxies be trusted?
SIP Security Mechanisms SIP is HTTP-like How do we secure HTTP services? –HTTP HTTPS (SSL) TSLTCP based Transport Layer Security (TSL) –E-mail PGP (Pretty Good Privacy) S/MIME –IP based communication IPsec (IP Security)
Solutions for securing SIP
HTTP Digest Authentication Example given in RFC 2617 –C–Client request (user: Mufasa passwd: Circle Of Life) –S–Server response: ?
Generating the MD5 values
HTTP Digest authentication parametermeaning realm client domain domain destination algorithm hash algorithm: MD5, MD5-sess nonce server-chosen nonce cnonce client-chosen nonce nc # times nonce has been used digest-uri destination qop protection (auth, auth-int) opaque string echoed by client username user ’ s name in specified realm response H(H(A1):nonce:nc:cnonce:qop:H(A2))
HTTP Digest authentication response = H(H(A1):nonce:nc:cnonce:qop:H(A2)) A1 = username:realm:password A2 = method:URI or method:URI:H(body) where H(x) = MD5(x)
SIP Proxy Digest Authentication Proxy Server using Digest Authentication Proxy Server Invite
SIP Proxy Digest Authentication Proxy Server using Digest Authentication Proxy Server Invite Challenge
TLS security: SIPS URI SIPS scheme added in RFC 3261 –sips:firstname.lastname@example.org TLS must be used on the whole path. Can not be applied to UDP-based SIP (only TCP or other reliable transport protocol) Applied hop-by-hop All SIP proxies required to implement
How to secure the talk? Securing the real-time media streams Multimedia streams are packet-oriented Encryptions and authentication algorithms should not cause too much delay Transmission must be UDP based Only two security mechanisms are currently available.
Securing the real-time media streams
Secure Real-Time Transport Protocol (SRTP) The Secure RTP Packet Format:
SRTP Default Encryption Algorithm
Secure Real-Time Transport Protocol (SRTP) The Secure RTCP Packet Format:
Conclusion VoIP security is complex –Numerous protocols –NAT/firewall traversal issues –QoS issues Technologies are in place to secure VoIP –Solutions we ’ ve discussed –However, no “ standard ” approach is being used Current VoIP providers do not secure calls
SIP follows HTTP programming model Three mechanisms suggested in IETF –Call Processing Language ( SIP – CPL ) –Common Gateway Interface ( SIP – CGI ) –SIP Servlet Other Options –Creation Markup Language (SCML) –Voice Extensible Markup Language (VoiceXML) –Call Control extensible Markup Language (CCXML)
SIP Programming Examples –“discard all calls from Monica during my business hours” –“redirect authenticated friends to my cell phone, anyone else to my secretary” –“if busy, return my homepage and redirect to recorder” Users and third parties may program
Where Services Locate? Source: H. Schulzrinne: “ Industrial Strength IP Telephony ”
Common Gateway Interface Almost identical to HTTP CGI Language independent ( Perl, Tcl, C, C++,... ) –Any binary may be executed as a separate program Communicates through IO and environment variables. –More flexible but more risky Unmanaged Resource Allocating –Single CGI may crash the server or user client Feb. 1, 2001: RFC 3050 (Common Gateway Interface for SIP) published
Call Processing Language Designed by the IETF to support sophisticated telephony services –May be used by both SIP or H.323. XML based scripting language –Extensive –Easily edited by GUI tools –Portability allows users to move across servers. Lightweight CPL interpreter is need –Better security
An Example A simple script that blocks anonymous callers
Java Servlets Similar to HTTP servlets Resource Managed By Container The class runs within a JVM (Java Virtual Machine) on server Security provided by Java Portable between OSs & servers
JAIN SIP The Java-standard interface to a SIP signaling stack. –Standardizes the interface to the stack. –Standardizes message interface. –Standardizes events and event semantics. –Application portability -verified via the TCK. Designed for developers who require powerful access to the SIP protocol. JAIN SIP can be utilized in a user agent, proxy, registrar or imbedded into a service container.
SIP Implementation Structure
Packages General package –Defines the architectural interfaces, the transaction and dialog interfaces and the event objects of the specification. Address package –Address package contains a generic URI wrapper and defines SIP URI and Tel URIs interfaces. Message package –Defines the interfaces necessary for the Request and Response messages. Header packages –Header package defines interfaces for all the supported headers and extension headers
HIGH-LEVEL SERVICE CREATION FRAMEWORK Service Creation Environment (SCE) –GUI Develop IDE Service Logic Execution Environment (SLEE)
HIGH-LEVEL SERVICE CREATION FRAMEWORK
Mechanism choosing Portability vs Performance –Portability needed if services deployed at multiple servers or end-devices. –Portable languages (CPL) need to be interpreted (processing delay) Deployment scenario decides service creation mechanism.
Implementations BaseVoice Vanilla –J2EE-based SIP Server, JAIN SIP API v1.1. SIPD –SIP CGI-BIN support Meetinghouse SIP Proxy –CPL support Source: “ http://www.iptel.org/info/products/ ”
Reference http://netlab.boun.edu.tr/mast/sip/ http://iptel.org/sip/siptutorial.pd f http://java.sun.com/products/jain/JAIN-SIP- Tutorial.pdf Creating Value Added Services in Internet Telephony: An Overview and a Case Study on a High-Level Service Creation Environment -- Roch H. Glitho, Ferhat Khendek, and Alessandro De Marco