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 Links to an external site..
- 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) Links to an external site.
- 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 Links to an external site., 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
- John Alexander (Editor), Chris Pearce, Anne Smith, Delon Whetten, Cisco CallManager Fundamentals: A Cisco AVVID Solution Links to an external site. Cisco Press, 2001, ISBN: 1-58705-008-0.
- Gonzalo Camarillo and Jonathan Rosenberg, SIP Demystified Links to an external site. McGraw-Hill Professional Publishing, 2001, ISBN: 0-07-137340-3.
- Daniel Collins, Carrier Grade Voice Over IP Links to an external site. McGraw-Hill Professional Publishing, 2000, ISBN: 0-07-136326-2.
- Jonathan Davidson, James Peters, Brian Gracely (Contributor), Jim Peters, Voice over IP Fundamentals Links to an external site., Cisco Press, 2000, ISBN: 1-5787-0168-6.
- Jonathan Davidson (Editor), Tina Fox (Editor), Phil Bailey (Editor)ConCon Deploying Cisco Voice Over IP Solutions Links to an external site., Cisco Press, 2001, ISBN: 1-58705-030-7.
- Bill Douskalis, Putting VoIP to Work: Softswitch Network Design and Testing Links to an external site., Prentice Hall, 2002, ISBN 0-13-040959-6.
- Bill Douskalis, IP Telephony: The Integration of Robust VoIP Services Links to an external site., Prentice Hall, 2000, ISBN 0-13-014118-6.
- Wenyu Jiang, Jonathan Lennox, Henning Schulzrinne and Kundan Singh, "Towards Junking the PBX: Deploying IP Telephony"
- Alan B. Johnston, SIP: Understanding the Session Initiation Protocol Links to an external site., Artech House, 2001, ISBN: 1-58053-168-7.
- Olivier Hersent, David, Gurle, and Jea-Pierre Petit, IP Telephony: Packet-based multimedia communication systems, Addison-Wesley, 2000, ISBN 0-201-61910-5.
- David Lovell and Scott Veibell Cisco IP Telephony Links to an external site., Cisco Press, 2001, ISBN: 1-58705-050-1.
- Mark A. Miller, Voice over IP Technologies: Building the Converged Network Links to an external site., Hungry Minds, Inc., 2002, ISBN 0764549073.
- Daniel Minoli, Delivering Voice over IP Networks Links to an external site., John Wiley and Sons, August 2002, ISBN 0-471-38606-5.
- David J. Wright, Voice over Packet Networks, John Wiley and Sons, 2001, ISBN 0-471-49516-6.
- The European Online Magazine for the IT Professional Links to an external site. http://www.upgrade-cepis.org Links to an external site. Vol. II, No. 3, Jun. 2001
- R.G. Cole and J.H. Rosenbluth, "Voice Over IP Performance Monitoring", Computer Communication Review, a publication of ACM SIGCOMM, volume 31, number 2, April 2001. ISSN # 0146-4833 is available from: http://www.acm.org/sigcomm/ccr/archive/2001/apr01/ccr-200104-cole.html Links to an external site.
- William C. Hardy, "VoIP Service Quality: Measuring and Evaluating Packet-Switched Voice", McGraw-Hill, January 2003, 317 pages, ISBN: 0071410767. (note the reviews Links to an external site. are very mixed on this book)
- Paul Mahler, VoIP Telephony with Asterisk Links to an external site., Signate, San Francisco, CA, 2004. ISBN 0-9759992-0-6
- Tom Tullis and Bill Albert, Measuring the User Expeience: Collecting, Analyzing, and Presenting Usability Metrics, Morgan Kaufmann, 2008, ISBN 978-0-12-373558-4
- Elisabetta Carrara, Security for IP multimedia applications over heterogeneous networks Links to an external site., Licentiate thesis, School of Information and Communication Technology (ICT), Microelectronics and Information Technology, IMIT), KTH Royal Institute of Technology, Trita-IMIT-LCN. AVH, ISSN 1651-4106; 05:01, 2005
- Moraitis Alexandros, IMS Security Links to an external site., M.Sc. thesis in Data Communication Systems, Brunel University, July 2010
Useful URLs
- J. Loughney and G. Camarillo, Authentication, Authorization, and Accounting Requirements for the Session Initiation Protocol (SIP) Links to an external site., RFC 3702, February 2004
- J. Rosenberg, A Session Initiation Protocol (SIP) Event Package for Registrations Links to an external site., RFC 3680, March 2004
- P. Faltstrom and M. Mealling, "The E.164 to Uniform Resource Identifiers (URI) Dynamic Delegation Discovery System (DDDS) Application (ENUM)" Links to an external site., RFC 3761, April 2004.
- J. Peterson, "enumservice registration for Session Initiation Protocol (SIP) Addresses-of-Record" Links to an external site., RFC 3764, April 2004
- O. Levin, "Telephone Number Mapping (ENUM) Service Registration for H.323" Links to an external site., RFC 3762, April 2004
- G. Kambourakis, D. Geneiatakis, S. Gritzalis, T. Dagiuklas, and C. Lambrinoudakis, Security and privacy issues towards ENUM protocol Links to an external site., in Signal Processing and Information Technology, 2005. Proceedings of the Fifth IEEE International Symposium on, 2005, pp. 478-483, DOI:10.1109/ISSPIT.2005.1577144.
- vovida.org Links to an external site. contains source code for the Vovida Open Copmmunication Application Library (VOCAL), which includes the servers described in the course book.
- note that Prof. H. Anthony Chan Links to an external site. of San Jose State University taugh a course "EE284 Convergent Voice and Data Network Links to an external site." during Fall 2002 that also used the same textbook as this course.
- Henning Schulzrinne's Session Initiation Protocol (SIP) web pages Links to an external site.
- Some SIP related Student Projects Links to an external site. done under the supervision of Prof. Henning Schulzrinne
- Columbia InterNet Extensible Multimedia Architecture CINEMA Links to an external site.
- IETF SIP Working group Links to an external site. (Note that this working group has concluded its work.)
- IP Telephony Links to an external site.
- SIP Forum
- SIP Center Links to an external site.
- Pulver.com Links to an external site. - see for example the May 02, 2010, Guest Blog by Daniel Berninger - Reboot VoIP by SIP Enabling Email Addresses Links to an external site.
- VoiceTronix Links to an external site. analog line cards
- Voxilla.org Links to an external site. hosts a collection of pointers to various open source telecom software projects for use with the GNU/Linux operating system
-
GNUComm
Links to an external site. pre-release versions of some GNUComm Components:
- GNU Bayonne Links to an external site., - Application Server -- a telecommunications application server; the focus is on voice response types of telephony applications.
- Babylon - Telephony Device Monitor
- TOSI - Client Call Control System
- Voice Mail - Multi-user messaging application
- Support Automation - Tele-support application
- Sales Automation - Tele-sales application
- NIST-SIP Links to an external site. a signaling stack and message parser for the SIP (Session Initiation Protocol); includes: a public domain extensible, modular JAVA based message parser for SIP, A simple stack with authentication, implementation of JAIN-SIP 1.0 interfaces, XML based call flow scripting tool, a test proxy with an XML interface for service creation, a trace viewer tool for visualization of message traces that passing through the stack
- J. van der Merwe, R. Cceres, Y-H. Chu, C. Sreenan. Mmdump - A Tool for Monitoring Internet Multimedia Traffic. ACM Computer Communication Review, 30(4), October 2000.http://citeseer.nj.nec.com/article/vandermerwe00mmdump.html Links to an external site.. See alsohttp://www.research.att.com/info/Projects/mmdump Links to an external site.
- C.J. Sreenan, Jyh-Cheng Chen, P Agrawal, and B Narendran, "Delay reduction techniques for playout buffering," IEEE Transactions on Multimedia, vol. 2, no. 2, June 2000.http://citeseer.nj.nec.com/sreenan00delay.html Links to an external site.
- End-to-End delay: http://wwwtvs.et.tudelft.nl/people/piet/papers/e2edelayripe_IEEE.pdf Links to an external site. see alsohttp://www.fokus.gmd.de/research/cc/glone/projects/cost263/meetings/09-namur/techdocs/Van-Mieghem-slides.pdf Links to an external site.
- Soonuk Seol and Myungchul Kim's Experiments and Analysis of VoIP over Mobile IP Links to an external site. slides - see the related PIMRC 2002 paper.
- Xianglin Deng, "Security of VoIP: Analysis, Testing and Mitigation of SIP-based DDoS attacks on VoIP Networks", Masters thesis, University of Canterbury, Canterbury, New Zealand, Available at http://ir.canterbury.ac.nz/handle/10092/2227, 2008.
- Xianglin Deng and Malcolm Shore, Advanced Flooding Attack on a SIP Server, presented at the International conference on Availability, Reliability and Security, 2009. ARES09., Fukuoka, 2009, pp. 647-651, DOI:10.1109/ARES.2009.15, Available at http://ieeexplore.ieee.org/lpdocs/epic03/wrapper.htm?arnumber=5066541.
- Stephen B. Wicker, "Cellular telephony and the question of privacy", Communications of the ACM, vol. 54, no. 7, p. 88, July 2011, DOI:10.1145/1965724.1965745, Available at http://portal.acm.org/citation.cfm?doid=1965724.1965745.
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 Links to an external site.
- example-call.tcpdump Links to an external site.
- 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.
- 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 Links to an external site., 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" Links to an external site., Official Gmail Blog, Wednesday, August 25, 2010 at 9:40 AM
- Google Talk Links to an external site. voice-chat
- Google's Gizmo5 Links to an external site.
- PeerMe Links to an external site., PeerMe, Inc.
- Yahoo! builds upon Dialpad acquisition to offer VoIP via its messanger Links to an external site.
- MCI Links to an external site. Web Calling for Windows Live Call
- Grandstream Networks Links to an external site.SIP 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 Links to an external site.
- 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 Links to an external site.
- tools Links to an external site. for testing your soundcard
- A useful tool for watching your SIP traffic is: ipgrab Links to an external site.
- A popular VoIP operator in the US is Vonage (http://www.vonage.com Links to an external site.)
- Jasomi Networks Links to an external site. recently annouced their PeerPoint Centrex Edition Links to an external site. device for serving VoIP customers behind NATs.
- Digisip Links to an external site. 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 Links to an external site. 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 Links to an external site.
- 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 Links to an external site..>
- OnDo's Brekeke Links to an external site. a commercial VoIP PBX and SIP server; with an emphasis on its web interface
- Digium Links to an external site. the primary developer and sponsor of Asterisk Links to an external site.™ is an open source linux based PBX
- minisip Links to an external site. - a SIP client with SRTP + MIKEY, developed by students from the course; see also the related eavesdropping tool "EVE"
- VoIPong Links to an external site. - utility which detects all Voice Over IP calls on a pipeline
- SJ Labs Links to an external site. SJphone - a SIP/H.323 softphone
- iptel.org's list of softphones Links to an external site.
- sipXphone Links to an external site.
- SIP express router Links to an external site.
- SIP Express Media Server Links to an external site. (SEMS)
- VOMIT Links to an external site. - voice over misconfigured internet telephones - given a tcpdump of a voice call creates a .wav file.
- INRIA Phoenix list of Links to an external site. SIP programs, testing, ...
- VoIP Security Workshop Links to an external site., June 1-2, 2005, Washington DC
- US National Institute of Standards and Technology(NIST), "Security Considerations for Voice Over IP Systems" Links to an external site., January 2005
- (U.S.) National Emergency Number Association (NENA), "NENA IP Capable PSAP Features And Capabilities Standard" Links to an external site., 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" Links to an external site., NENA-08-501, June, 2004
- AudioCodes Links to an external site. VoIP, especially voice compression technology
- "Connexion by Boeing" Links to an external site. - be on-line even from aircraft
- VoP Security Forum Links to an external site. has a tool: SiVuS Links to an external site. - The VoIP Vulnerability Scanner
- Blue Box Podcast #22: SIP Security at IETF (part 1), VoIP security news, comments and more Links to an external site., 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
Links to an external site.(CONFerencing IMS-enabled Architecture for Next-generation Communication Experience).
For additional details about the proctocols see the IETF Centralized Conferencing (XCON) Links to an external site. working group. - Alberto Escudero Pascual and Louise Berthilson, "VoIP-4D Primer- Building Voice Infrastructure in Developing Regions Links to an external site., Translators: Anas Tawileh (Arabic), Johan Bilien (French). Available in English, Arabic, French, and Spanish.
- SIP debugging and testing
- Sipp (sipp.sourceforge.net Links to an external site.) - conformance testing tool
- SIP Swiss Army Knife (SIP SAK Links to an external site.) - 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 Links to an external site., 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 Links to an external site.", 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 Links to an external site.", IETF, Network Working Group, Request for Comments: 5638, September 2009
- Shanbo Li's masters thesis ( mirror Links to an external site.) "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 Links to an external site.
- minisip with Ubuntu 11.10:
- sudo apt-get update
- sudo apt-get install build-essential
- sudo apt-get install libglademm-2.4-dev
- sudo apt-get install libssl-dev
- As per http://lists.minisip.org/pipermail/minisip-users/2007-February/001297.html
Links to an external site.
echo /usr/local/share/aclocal >> /usr/share/aclocal/dirlist - Then build as per the instructions at http://www.minisip.org/develop_build.html Links to an external site.http://www.minisip.org/develop_build.html
- 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 - 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? Links to an external site., WhichVoip.com - describes a number of IP PBXs.
Acronyms and Abbreviations useful for IK2554
Acronym or Abbreviation | Description | Swedish 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 | |
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
Videos
- Video part 1 - 20150909 (revised)
- Video part 2 - 20150911
- Video part 3 - 20150911
- Video part 4 - 20150911