2015

Aim

This course will give both practical and general knowledge concerning Voice over IP. The emphasis will be on the underlying protocols.

Learning Outcomes

Following this course a student should be able to:

  • Understand the relevant protocols (particularly SIP, SDP, RTP, and SRTP): what they are, how they can be used, and how they can be extended.
  • Enable you to utilize SIP in Presence and event-based communications
  • Understand how SIP can provide application-level mobility along with other forms of mobility
  • Understand how SIP can be used to facilitate communications access for users with disabilities (for example using real-time text, text-to-speech, and speech-to-text) and to know what the basic requirements are to provide such services
  • Understand SIP can be used as part of Internet-based emergency services and to know what the basic requirements are to provide such services
  • Contrast "peer-to-peer" voice over IP systems (i.e., how they differ, how they might scale, what are the peers, ...)
  • Know the relevant standards and specifications - both of the protocols and of the requirements (for example, concerning legal interception)
  • Understand the key issues regarding quality-of-service and security
  • Evaluate existing voice over IP and other related services (including presence, mobile presence, location-aware, context-aware, and other service)
  • Design and evaluate new SIP based services
  • Read the current literature at the level of conference papers in this area.

While you may not be able to understand all of the papers in journals, magazines, and conferences in this area - you should be able to read 90% or more of them and have good comprehension. In this area it is especially important that develop a habit of reading the journals, trade papers, etc. In addition, you should also be aware of both standardization activities, new products/services, and public policy in the area.

  • Demonstrate knowledge of this area both orally and in writing.

By writing a paper suitable for submission to conferences and journals in the area.

    This course should prepare you for starting a thesis project in this area (for undergraduate students) or beginning a thesis or dissertation (for graduate students).

    Contents

    This course will focus on the protocols associate with Voice over IP. The course should give both practical and more general knowledge concerning the these protocols. One of the major aims of the course is that student should be able to build upon these protocols to enable new services.

    Topics

    • Session Initiation Protocol (SIP)
    • Real-time Transport Protocol (RTP)
    • Real-time Streaming Protocol (RTSP)
    • Common Open Policy Server (COPS)
    • SIP User Agents
    • Location Server, Redirect Server, SIP Proxy Server, Registrar Server, ... , Provisioning Server, Feature Server
    • Call Processing Language (CPL)

    Examination Requirements

    • An assigned paper requiring roughly 50-200h of work by each student (7.5 ECTS)
    • Registration: Monday 28 September 2015 at 23:59, to maguire@kth.se with the subject: "IK2554 topic" giving:
      • Group members, leader.
      • Topic selected
    • Written report
      • The length of the final report should be 10 pages (roughly 5,000 words) for each student; it should not be longer than 12 pages for each student - papers which are longer than 12 pages per student will be graded as "F".
      • If there are multiple students in a project group, the report may be in the form of a collections of papers, with each paper suitable for submission to a conference or journal.
      • Contribution by each member of the group - must be clear (in the case where the report is a collection of papers - the role of each member of the group can be explained in the overall introduction to the papers.
      • The report should clearly describe: 1) what you have done; 2) who did what; if you have done some implementation and measurements you should describe the methods and tools used, along with the test or implementation results, and your analysis.
      • Final Report: written report due Friday 23 October 2015 + oral presentations individually scheduled during week 44 (26-30 October 2015).
      • Send email with URL link to maguire@kth.se and you must submit your Zotero RDF file or BibTeX file of references that you have used.
      • Late assignments will not be accepted (i.e., there is not guarantee that they will be graded before the end of the term)
      • Note that it is permissible to start working well in advance of the deadlines!
      • For graduate students the paper should be of the quality that it could be submitted to a conference - immediately following the course.
    • Oral presentations; Each group should present their results in 15 minutes or less. The presentation should not be more than 15 minutes, given the 20 minute time-slot this gives time for a couple questions and changing presenters. You only need to attend the part of the day that you give your presentation.

    Code of Honor and Regulations

    It is KTH policy that there is zero tolerance for cheating, plagiarism, etc. - for details see KTH Student's rights and obligations See also the KTH Ethics Policies

    Grades

    For new ECTS grading:

    • To get an "A" you need to write an outstanding or excellent paper and give an outstanding or excellent oral presentation. (Note that at least one of these needs to be excellent.)
    • To get a "B" you need to write a very good paper, i.e., it should be either a very good review or present a new idea; and you have to give a very good oral presentation.
    • To get a "C" you need to write a paper which shows that you understand the basic ideas underlying voice over IP and that you understand one (or more) particular aspects at the level of an average masters student. In addtion, you must be able to present the results of your paper in a clear, concise, and professional manner - and answer questions (as would be expected at a typical international conference in this area.)
    • To get a "D" you need to demonstrate that you understand the basic ideas underlying voice over IP, however, your depth of knowledge is shallow and you are unable to orally answer indepth questions on the topic of your paper.
    • If your paper has some errors (including incomplete references) or you are unable to answer any indepth questions following your oral presentation the grade will be an "E".
    • If your paper has serious errors or you are unable to answer basic questions following your oral presentation the grade will be an "F".
    • If your paper or oral presentation are close to passing, but not at the passing level, then you will be offered the opportunity for "komplettering", i.e., students whose written paper does not pass can submit a revised version of their paper (or a completely new paper) - which will be evaluated; similarly students whose oral presentation is unacceptable may be offered a second opportunity to give their oral presentation. If a student fails the second oral presentation, they must submit a new paper on a new topic in order to give an oral presentation on this new topic.

    Suggestions when writing your report

    Suggestions when writing your report

    • For students who are looking for examples of papers - sFor students who are looking for examples of papers - see the ACM Sigcomm proceedings.
    • For your document, you should be sure to use A4 sized paper rather than US letter.
    • For those using LaTeX, you can improve the look of the document by:
      • switching to using PostScipt fonts (instructions)
      • You can also turn off hyphenation or at least limit its use with "\hyphenpenalty=5000 \tolerance=1000"

    Some common flaws in reports (from other courses)

    • Using the KTH logotype on your report or presentation is not permitted.
    • Incomplete references
    • Missing important citations
    • Statements made without justification or supporting citations
    • Poor (or no) editing
    • Failure to spell check the document
    • Documents which it is clear that no one looked at after formatting - often these have breaks in the middle of sentences, missing phrases, ... .
    • Using figures from others without the copyright owner's permission
    • Lack of page numbers
    • Unreadable text in figures
    • Failure to label elements of figures adequately
    • Use of contractions - you should avoid using contractions in a formal report
    • Use of acronyms or abbreviations without properly introducing them; often failure to use these acroynms and abbreviations consistently through the rest of the paper
    • Redundant text
    • Using too few refences, often the paper looks like simply a cut and paste edit of these references.
    • Single sentence paragraphs
    • Lack of vertical white space between paragraphs, which in some cases makes it hard to understand where new paragraphs begin
    • Lack of a date - every document should have a date, in addition to title and author(s), and most references should have more information - keep in mind that the goal is to enable someone (perhaps even yourself) to find the reference at a later point in time
    • Sections and subsections are not numbered - hiding both the structure of the document and making cross references difficult

    Literature

    Main Textbook

    The course will mainly be based on the book: Henry Sinnreich and Alan B. Johnston, Internet Communications Using SIP: Delivering VoIP and Multimedia Services with Session Initiation Protocol, 2nd Edition, Wiley, August 2006, ISBN: 0-471-77657-2

    Additional Reference Books

    none - at the present time

    Lecture notes are available on-line in PDF format. See the notes associated with each of the course topics.

    Errata for Henry Sinnreich and Alan B. Johnston, Internet Communications Using SIP: Delivering VoIP and Multimedia Services with Session Initiation Protocol (note this is a work in progress)

    Lecture notes

    Revised verson of lecture notes for Fall 2015 (~22MB)

    Note that this version of the notes is searchable - even the text in each slide is text!

    Course election presentation

    Slides presented on 2014.05.15 for students electing their fall courses.

    Supplementary readings

    Useful URLs

    Other on-line Course related Material

    • A sample call and how to record with tcpdump and decode with tcpdump, ethereal, and ipgrab.
      • Running /usr/local/vocal/bin/sipset as user 1010 on a linux PC named "tlclab01" (which will have the SIP URL sip:1010@192.168.194.24) and making a call to 1010@172.18.194.18 (which will have the SIP URL sip:1010@172.18.194.18). Thus user 1010 on tlclab01 makes a call, which user 1010 on 172.18.194.18 (a Cisco ATA 186) answers.
        At the end of the call, the user on tlclab01 hangs up.
      • SIP-call-example
      • rtp-filter.ethereal
      • example-call.tcpdump
    • An example paper from 2011: Javier Lara Peinado and Víctor Ariño Pérez, Context-aware VoIP
    • Examples of written reports submitted in 2004:
      Andreas Ångström and Johan Sverin, VoiceXML and Khurram Jahangir Khan and Ming-Shuang Lang, Voice over Wireless LAN and analysis of MiniSIP as an 802.11 Phone both reports appear here with permission of the authors.
    • Stefano Ventura, VoIP & Security for Enterprise, Séminaire Grifes, Institute of Information and Communication Technologies (IICT), Haute Ecole d'Ingénierie et de Gestion du Canton de Vaud (HEIG-VD), Yverdon Switzerland, 8.11.2005 - a very nice introduction to VoIP security (in french)


    Sources for Further Information

    • The course previously used: Henry Sinnreich and Alan B. Johnston, Internet Communications Using SIP: Delivering VoIP and Multimedia Services with Session Initiation Protocol, Wiley, 2001, ISBN: 0-471-41399-2 and a second book Luan Dang, Cullen Jennings, and David Kelly, Practical VoIP: Using VOCAL, O'Reilly, 2002, ISBN 0-596-00078-2.
    • Robin Schriebman, "Call phones from Gmail", Official Gmail Blog, Wednesday, August 25, 2010 at 9:40 AM
    • Google Talk voice-chat
    • Google's Gizmo5
    • PeerMe, PeerMe, Inc.
    • Yahoo! builds upon Dialpad acquisition to offer VoIP via its messanger
    • MCI Web Calling for Windows Live Call
    • Grandstream NetworksSIP phones and analog telephone adpators
    • Internet Voice Campaign - part of the Voice On the Net (VON) Coalition (www.von.org)
      Founding members of the Internet Voice Campaign include EarthLink, Google, Level 3, Pulver.com, Skype, Sonus Networks, and USA Datanet.
    • How to use SER with CPL
    • To use SER with TLS and minisip - change your TLS method in the openser.conf file to: "tls_method = SSLv23" (thanks to Pjothi's comments in the minisip users mailing list on Fri, 3 Feb 2006
    • thevoipweblog
    • tools for testing your soundcard
    • A useful tool for watching your SIP traffic is: ipgrab
    • A popular VoIP operator in the US is Vonage (http://www.vonage.com)
    • Jasomi Networks recently annouced their PeerPoint Centrex Edition device for serving VoIP customers behind NATs.
    • Digisip offers flat rate pricing to the swedish fixed network for 195 SEK/month {seems to be limited to 30 hours}
    • Bredbandsbolaget offers per minute pricing to the swedish fixed and mobile networks.
    • See the excellent list of references which Raj Jain has made available
    • Christian Hoene and Enhtuya Dulamsuren-Lalla of TU-Berlin, TKN, have developed a really nice application for showing the effect of packets loss on VoIP quality - Mongolia: An Auditory Testing Environment to Study the Importance of a VoIP Packet
    • For access to the KTH electronic library see KTHB e-library.
    • Texas A&M University (TAMU) and Internet2 have created a Internet2 Technology Evaluation Center (ITEC) focused on Voice over IP.>
    • OnDo's Brekeke a commercial VoIP PBX and SIP server; with an emphasis on its web interface
    • Digium the primary developer and sponsor of Asterisk™ is an open source linux based PBX
    • minisip - a SIP client with SRTP + MIKEY, developed by students from the course; see also the related eavesdropping tool "EVE"
    • VoIPong - utility which detects all Voice Over IP calls on a pipeline
    • SJ Labs SJphone - a SIP/H.323 softphone
    • iptel.org's list of softphones
    • sipXphone
    • SIP express router
    • SIP Express Media Server (SEMS)
    • VOMIT - voice over misconfigured internet telephones - given a tcpdump of a voice call creates a .wav file.
    • INRIA Phoenix list of SIP programs, testing, ...
    • VoIP Security Workshop, June 1-2, 2005, Washington DC
    • US National Institute of Standards and Technology(NIST), "Security Considerations for Voice Over IP Systems", January 2005
    • (U.S.) National Emergency Number Association (NENA), "NENA IP Capable PSAP Features And Capabilities Standard", Document 58-001, Arlington, VA, February 1, 2005.
    • (U.S.) National Emergency Number Association (NENA) Migration Working Group of the Network Technical Committee, "NENA Technical Information Document on the Network Interface to IP Capable PSAP", NENA-08-501, June, 2004
    • AudioCodes VoIP, especially voice compression technology
    • "Connexion by Boeing" - be on-line even from aircraft
    • VoP Security Forum has a tool: SiVuS - The VoIP Vulnerability Scanner
    • Blue Box Podcast #22: SIP Security at IETF (part 1), VoIP security news, comments and more, April 7, 2006
    • University of Naples, in cooperation with Ericsson Nomadic Lab in Helsinki, have released a first implementation of an XCON-compliant conferencing platform. The server side is based on Asterisk and a modified version of its MeetMe application, while the client side is based on Minisip. The system uses the Binary Floor Control Protocol (BFCP). The project is called CONFIANCE(CONFerencing IMS-enabled Architecture for Next-generation Communication Experience).
      For additional details about the proctocols see the IETF Centralized Conferencing (XCON) working group.
    • Alberto Escudero Pascual and Louise Berthilson, "VoIP-4D Primer- Building Voice Infrastructure in Developing Regions, Translators: Anas Tawileh (Arabic), Johan Bilien (French). Available in English, Arabic, French, and Spanish.
    • SIP debugging and testing
      • Sipp (sipp.sourceforge.net) - conformance testing tool
      • SIP Swiss Army Knife (SIP SAK) - a useful command line tool for SIP development and administration
    • Michael Vorländer, "Auralization of spaces", Physics Today, Volume 62 (6), June 2009, pp. 35-40.
    • For some figures on power consumption of IP phones see Green-VoIP with snom, based on the study: "Leistungsmessung von IP-Phones", FH Frankfurt am Main - University of Applied Sciences, Forschungsgruppe für Telekommunikationsnetzte, Dezember 2008
    • J. Rosenberg, "A Hitchhikers' Guide to SIP", IETF, Network Working Group, RFC 5411, January 2009
    • H. Sinnreich (Ed.), A. Johnston, E. Shim, and K. Singh, " Simple SIP Usage Scenario for Applications in the Endpoints", IETF, Network Working Group, Request for Comments: 5638, September 2009
    • Shanbo Li's masters thesis ( mirror) "Web Call Application" (August 2009) illustrates how SIP can be used to instantiate calls between parties by using a web server. It provides a good example of how third party call control can be used.
    • Adeel Ahmed, Habib Madani, and Talal Siddiqui. VoIP performance management and optimization:A KPI-based approach to managing and optimizing VoIP networks. Indianapolis, IN: Cisco Press, 2011, 448 pages, ISBN-13: 978-1-58705-528-7.
    • Converting a PCAP file to LaTeX can be done with pcap2tex
    • minisip with Ubuntu 11.10:
      1. sudo apt-get update
      2. sudo apt-get install build-essential
      3. sudo apt-get install libglademm-2.4-dev
      4. sudo apt-get install libssl-dev
      5. As per http://lists.minisip.org/pipermail/minisip-users/2007-February/001297.html
        echo /usr/local/share/aclocal >> /usr/share/aclocal/dirlist
      6. Then build as per the instructions at http://www.minisip.org/develop_build.htmlhttp://www.minisip.org/develop_build.html
      7. I built the files under /home/maguire/trunk, thus I added:
        /usr/local/lib/libminisip/plugins
        /home/maguire/trunk/install/x86-pc-linux-gnu/usr/lib
        to the file /etc/ld.so.conf.d/i386-linux-gnu_GL.conf
      8. I also defined LD_LIBRARY_PATH to include /usr/local/lib in addition to /usr/lib and /lib
    • for patterns of communication between different speakers, see the book: Deborah Tannen, You Just Don't Understand: Women and Men in Conversation, William Morrow Paperbacks, 352 pages, ISBN-10: 0060959622, ISBN-13: 978-0060959623, July 24, 2001
    • Tony Campbell, IP PBX Phone Systems and Providers: What is an IP enabled PBX?, WhichVoip.com - describes a number of IP PBXs.

    Acronyms and Abbreviations useful for IK2554

    Acronym or AbbreviationDescriptionSwedish term
    2B+D 2 B channels and one D channel (an ISDN line)
    30B+D 30 B channels and one D channel (an E1 as ISDN)
    3D Three Dimensional
    3G Third Generation
    3GPP 3rd Generation Partnership Project
    3GPP2 3rd Generation Partnership Project 2
    4ESS #4 Electronic Switching System
    5ESS #5 Electronic Switching System
    AAA Authentication, Authorization, and Accounting
    AAB Automatic Alternative Billing
    ABD Abbreviated dialing
    ACB Automatic Callback
    ACC Account Card Calling
    ACD Automatic Call Distribution
    ACK Acknowledgment
    ACM Association for Computing Machinery
    AES Advanced Encryption Standard
    AIN Advanced Intelligent Network
    ALG Application Level Gateway
    AM Ante meridiem (Latin for Before noon)
    AN Access Network
    ANAT Alternative Network Address Types
    ANM Answer message
    AOR Address of Record
    AP Access Point
    API Application Programming Interface

    app

    Application app
    APS Application Policy Server
    ARM Advanced RISC Machine
    ARP Address Resolution Protocol
    AS Application Server
    ATA Analog Telephone Adapter
    ATM Asynchronous Transfer Mode
    ATT Attendant
    AUTC Authentication
    AUTZ Authorization code
    AV Audio Visual
    AVATS AccessGrid Video and Audio Tools Support
    AVH avhandling (Swedish)
    AXE Automated Telephone Exchange (a series of exchanges)
    B2BUA Back-to-Back User Agent
    BCAST Broadcast
    BCP Best Current Practice
    BER Bit Error Rate
    BFCP Binary Floor Control Protocol
    BGCF Breakout Gateway Control Function
    BGP Border Gateway Protocol
    BHCA Busy Hour Call Attempts
    BIND Berkeley Internet Name Domain (DNS server software)
    BT Block Type
    CA Certificate Authority
    CALEA Communications Assistance for Law Enforcement Act
    CB Citizen Band (radio)
    CBC Cipher Block Chaining
    CC Country Code
    CCBS Completion of calls to busy subscriber
    CCC Credit Card Calling
    CCITT Consultative Committee on International Telephone and Telegraph
    CCNC Consumer Communications and Networking Conference
    CCXML Call Control Extensible Markup Language
    CCXML4J CCXML Interpreter in Java
    CD Call Distribution
    CDC Conference on Decision and Control
    CDP Cisco Discovery Protocol
    CDR Call Detail Record
    CELEX Communitatis Europeae Lex (European Union law database)
    CELP Codetopic-excited linear predictive
    CEST Central European Summer Time
    CF Call Forwarding
    CFC Call forwarding on busy/don’t answer
    CGI Common Gateway Interface
    CHA Call hold with announcement
    CIO Chief Information Officer
    CLF Common Log Format
    CLUE ControLling mUltiple streams for tElepresence
    CNAME Canonical Name
    COC Consultation calling
    CODEC Coder/Decoder kodek
    COMET SIP Preconditions Met message
    CON Conference calling
    COPS Common Open Policy Service
    CPE Customer Premises Equipment
    CPL Call Processing Language
    CPM Customer Profile Management
    CPNI Customer Proprietary Network Information
    CPU Central Processing Unit
    CRD Call Rerouting Distribution
    CRG Customized ringing
    CS Computer Science
    CSI Computer Security Institute
    CTI Computer Telephony Integration
    CUCS Columbia University Computer Science (Department)
    CUG Closed user group
    CW Call Waiting
    DAA Data Access Arrangement
    DB Database
    DCON Distributed Conferencing
    DCR Destination Call Routing
    DCS Distributed Call Signaling
    DDDS Dynamic Delegation Discovery System
    DDS Delegation Discovery System
    DEA (U.S.) Drug Enforcement Administration
    DEFSOP Digital Evidence Forensics Standard Operating Procedures
    DES Data Encryption Standard
    DHCP Dynamic Host Configuration Protocol
    DHCPACK Dynamic Host Configuration Protocol Acknowledgment
    DMP Device Message Protocol
    DNR Diarienummer (Swedish)
    DNS Domain Name Server
    DOI Digital Object Identifier
    DOJ (U.S.) Department of Justice
    DOM Document Object Model
    DOP Default Outbound Proxy
    DS0 Digital Signal 0 (a level in the Digital signaling hierarchy)
    DSL digital subscriber line
    DTD Document Type Definition
    DTMF Dual Tone Multi-Frequency
    DUP Destinating User Prompter
    E1 2 Mbps time-multiplexed link
    E911 Enhanced 911 (emergency service)
    EDS Electronic Data Systems Corporation
    ENUM E-number
    ENUMWG ENUM Working Group
    ESP Enhanced Service Provider
    ESR Emergency Services Routing
    ETNS European Telephony Numbering Space
    ETS Emergency Telecommunications Service
    ETSI European Telecommunications Standards Institute
    EU European Union
    EUDRD EU Data Retention Directive
    FAQ Frequently Ask Questions
    FAX Facsimile
    FBI (U.S.) Federal Bureau of Investigation
    FCC (U.S.) Federal Communications Commission
    FCP Firewall Communication Protocol
    FEC Forward Error Correction
    FID Flow Identification
    FISA (U.S.) Foreign Intelligence Surveillance Act
    FLUTE File Delivery over Unidirectional Transport
    FMD Follow Me Diversion
    FPH Freephone
    FQDN Fully Qualified Domain Name
    FRA Försvarets Radioanstalt (Swedish for Defense Radio Institute)
    FS Feature Server
    FXO Foreign eXchange Office
    FXS Foreign eXchange Subscriber (FXS)
    GAA Generic Authentication Architecture
    GAP Call gapping
    GBA Generic Bootstrapping Architecture
    GDS Global Dialing Scheme
    GEOPRIV Geographic Location/Privacy
    GLP Gateway Location Protocol
    GMT Greenwich Mean Time
    GNOME GNU Network Object Model Environment
    GPL GNU Public License
    GPRS General Packet Radio Service
    GRUU Globally Routable User Agent (UA) URI
    GSM Global System for Mobile telecommunications (original name in French: Groupe Spécial Mobile)
    GSTN Global Switched Telephone Network
    HCI Human-Computer Interface
    HERFP Heterogeneous Error Response Forking Problem
    HF Human Factors
    HIP Host Identity Protocol
    HMAC Hashed Message Authentication Code
    HNT Hosted Nat Traversal
    HSS Home Subscriber Server
    HTML Hypertext Markup Language
    HTTP Hypertext Transfer Protocol
    IA Information Appliance
    IAA I Am Alive
    IAB Internet Architecture Board
    IAM Initial Address Message
    IANA Internet Assigned Number Authority
    IAP Intercept Access Point
    IAX Inter-Asterisk eXchange
    IAX2 Inter-Asterisk eXchange version 2
    IC Integrated Circuit
    ICC (IEEE) International Conference on Communications
    ICE interactive connectivity establishment
    ICE interactive connectivity establishment
    ICMP Internet Control Message Protocol
    ICON International Conference on Networks
    ICT Information and Communications Technology
    ICW internet call waiting
    ID identifier
    IDDD International Direct Distance Dialing
    IEEE Institute of Electrical and Electronics Engineers
    IETF Internet Engineering Task Force
    IFF if and only if
    IKE Internet Key Exchange
    IM IP Multimedia
    IMAC Innovative Mobile Applications of Context
    IMS IP Multimedia Subsystem
    IN Intelligent Networks or IP Networks
    INET Internet Society (Conference series)
    info information
    IOS (Cisco) Internetwork Operating System
    IP Internet Protocol
    IP IM Presence
    ISBN International Standard Book Number
    ISDN Integrated Services Digital Network
    ISO International Standards Organization
    ISP Internet Service Provider
    ISSN International Standard Serial Number
    ISUP Integrated Services Digital Network (ISDN) User Part
    IT Information Technology
    ITAD Internet Telephony Administrative Domain
    ITU International Telecommunication Union
    IVR Interactive voice response
    JAIN Java APIs for Integrated Networks
    JB Jitter Buffer
    JBA Jitter Buffer Adaptive
    JDSU JDS Uniphase Corporation
    JSR Java Specification Requests
    JVM Java Virtual Machine
    KB Knowledge base
    KDE K Desktop Environment
    KPI Key Performance Indicators
    KTH Kungliga Tekniska Högskolan (KTH Royal Institute of Technology)
    L16 Linear 16 bit (encoding)
    L8 Linear 8 bit (encoding)
    LA Los Angeles (California)
    LAN Local Area Network
    LCD Liquid Crystal Display
    LDAP Lightweight Directory Access Protocol
    LEA Law Enforcement Administration
    LGPL GNU Lesser General Public License
    LI Lawful Interception
    LIM Call limiter
    LOG Call logging
    LPC Linear Predictive Coding
    LRF Location Retrieval Function
    LS Location Server
    LSI Large Scale Integration
    M2M Machine-to-Machine
    MAC Medium Access Control (below the link layer) or message authentication code
    MAP Mobility Anchor Point or Mobile Application Part
    MAS Mass calling
    max maximum
    MB Megabyte
    MCI Malicious Call Identification
    MDC Modification Detection Code
    MEST Middle European Summer Time
    MGCF Media Gateway Control Function
    MGCP Media Gateway Control Protocol
    MGW Media Gateway
    MIB Management Information Base
    MIDCOM Middlebox Communications
    MIKEY Multimedia Internet KEYing
    MIME Multipurpose Internet Mail Extensions
    min minimum
    MMC Meet Me Conference
    MMUSIC Multiparty Multimedia Session Control
    MOS Mean Opinion Score
    MPA MPEG Audio
    MPEG Motion Picture Experts Group
    MPLS Multiprotocol Label Switching
    MS Master's of Science or Marshal Server
    ms millisecond
    MSCML Media Server Control Markup Language
    MSML Media Server Markup Language
    MUST Militära Underrättelse- och Säkerhetstjänsten (Swedish for Military Intelligence and Security Service)
    MWC Multi Way Calling
    NAPTR Naming Authority Pointer
    NAT Network Address Translator
    NB Nota bene (Latin meaning Note well)
    NENA National Emergency Number Association
    NIST National Institute of Standards and Technology
    NJ New Jersey
    NOSSDAV Network and Operating System Support for Digital Audio and Video
    NP Number Portability
    NSA (U.S.) National Security Agency
    NSIS Next Steps In Signaling
    NTP Network Time Protocol
    NY New York
    OCHA (United Nations) Office for Coordination of Humanitarian Affairs
    OCS Originating Call Screening
    ODR Origin Dependent Routing
    OK Okay
    OMA Open Mobile Alliance
    ONC Off Net Calling
    ONE One number
    OR Oregon
    OSP Open Settlement Protocol
    OUP Originating User Prompter
    PAMS Perceptual Analysis Measurement System
    PBX Private Branch Exchange
    PC Personal Computer
    PCC Probabilistic Congestion Control
    PCM Pulse Code Modulation
    PCMU Pulse Coded Modulation mu-law (encoding)
    PDA Personal Digital Appliance
    PDF Portable Document Format
    PESQ Perceptual Evaluation Speech Quality
    PIDF Presence Information Data Format
    PINT PSTN Internet Internetworking
    PLC Packet Loss Concealment
    PLS Pronunciation Lexicon Specification
    PM Post meridiem (Latin for afternoon)
    PN Personal Numbering
    PNP Private Numbering Plan
    POP Point of Presence
    POTS Plain Old Telephony Service
    PRACK Reliable ACK
    PRIV a private extension to SDES
    PRM Premium rate
    PRMC Premium charging
    PS Provisioning Server
    PSAP public safety answering point
    PST Pacific Standard Time
    PSTN Public Switched Telephone Network
    PT Payload Type
    PTS Post- och Telestyrelsen (Swedish)
    Q3 Network Management Interface (ITU-T)
    QCLEP Qualcomm Code Excited Linear Prediction
    QR Quick Response (Code)
    QSIG Q-Signaling protocol
    RADIUS Remote Authentication Dial-In User Service
    RAM Random Access Memory
    RAT Robust Audio Tool
    RC Report Count
    RDF Resource Description Framework
    RE regarding
    REC recommendation
    REVC Reverse charging
    RFC Request for Comments
    RG Residential Gateway
    RIA Rich Internet Application
    RIPE Reseaux IP Europeens
    RISC Reduce Instruction Set Computer
    RMT Reliable Multicast Transport
    RR Receiver Report
    RS Receives SIP
    RSS Rich Site Summary
    RST reset
    RSVP Resource Reservation Protocol
    RTCP Real time Control Protocol
    RTCP Real Time Control Protocol
    RTP Real Time Protocol
    RTSP Real Time Streaming Protocol
    RTT Round Trip Time
    RX Receiver
    SA Société Anonyme (French for Corporation)
    SAP Session Announcement Protocol
    SCCP Skinny Client Control Protocol
    SCF Selective Call Forwarding
    SCTP Stream Control Transmission Protocol
    SDCS Security in Distributed Computing Systems
    SDES Source Description
    SDK Software Development Kit
    SDP Session Description Protocol
    SE Sweden
    SEC Security screening
    SER SIP Express Router
    SFS Svensk Författningssamling (Swedish for Swedish Code of Statutes)
    SIGAD SIGINT Activity Designator
    SIGINT Signals Intelligence
    SIMPLE SIP Instant Messaging Presence Leveraging Extensions
    SIP Session Initiation Protocol
    SIPAPI SIP API
    SIPIT Session Initiation Protocol Interoperability Test
    SIPPING Session Initiation Protocol Project INvestiGation
    SIPREC SIP-Based Media Recording
    SIPUA SIP User Agent
    SLA Service Level Agreement
    SLF Subscription Locator Function
    SLP Service Location Protocol
    SMIL Synchronized Multimedia Integration Language
    SMS Short Message Service
    SMTP Simple Mail Transfer Protocol
    SNMP Simple Network Management Protocol
    SOAP Simple Object Access Protocol
    SPA Service Provider APIs
    spam unsolicited bulk messages
    SPIRITS Service in the PSTN/IN Requesting InTernet Service
    SPIT Spam over Internet Telephony
    SPL Split charging
    SPLC Split charging
    SR Sender Report
    SRGS Speech Recognition Grammar Specification
    SRTP Secure Real Time Protocol
    SSM SIP Service Manager
    SSML Speech Synthesis Markup Language
    SSO Single Sign-On
    SSRC Synchronization Source
    STUN Session Traversal Utilities for NAT
    TCP Transmission Control Protocol
    TCS Terminating Call Screening
    TDM Time Division Multiplexing
    telecom telecommunications
    TESLA Timed Efficient Stream Loss-Tolerant Authentication
    TFRC TCP-Friendly Rate Control
    TFWC TCP-Friendly Window-based Congestion Control
    TGREP Telephony Gateway REgistration Protocol
    TI Texas Instruments
    TIPHON Telecommunications Internet Protocol Harmonization Over Networks
    TLS Transport Layer Security
    TMN Telecommunications Management Network
    TOS Type of Service
    TR Technical Report
    TRA Call transfer
    TRIP Telephony Routing over IP
    TS Technical Specification
    TTL Time to live
    TURN traversal using relay nat
    UA User Agent
    UAC User Agent Client
    UAN Universal Access Number
    UAS User Agent Server
    UC Unified Communications
    UCL University College London
    UCS Universal Character Set
    UDP User Datagram Protocol
    UDR User Defined Routing
    UK United Kingdom
    UMA Unlicensed Mobile Access
    UML Universal Modeling Language
    UN United Nations
    UPT Universal Personal Telecommunication
    URI Uniform Resource Identifier
    URL Universal Resource Locator webbadress
    URN Uniform Resource Name
    US United States (of America)
    USB Universal Serial Bus
    USC United States Code
    USR User
    UTF UCS Transformation Format
    VA Virginia
    VAD Voice Activity Detection
    VB Visual Basic
    VDSL Very high speed Digital Subscriber Line
    VER Version
    VLAN virtual local area network
    VoIP Voice over IP
    VOT Televoting
    VP Vice Preseident
    VPN virtual private network
    VQT Voice Quality Tester
    VXML Voice XML
    VoIP Voice over IP
    W3C World Wide Web Consortium
    WAN Wide Area Network
    WG Working Group
    WLAN Wireless Local Area Network
    XML Extensible Markup Language
    XMPP Extensible Messaging and Presence Protocol
    ZRTP (Phil Zimmermann) Z Real-time Transport Protocol

    Vocabulary from Lectures (English roots and possible Swedish equivalents)

    IK2554 vocabulary list - 2014.08.18


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