VT 2005, Period 4, 2G1325 and 2G5564 Practical Voice Over IP (VoIP): SIP and related protocols
(Röst över IP (VoIP) i praktiken: SIP och relaterade protokoll)

Announcements

NOTE

List of papers for 3 June (note they will be intermixed with papers from 2G1330

• Everyone who has submitted a report - should have been assigned a scheduled time for their oral presental - if you do not have an acknowledgement of your report or a scheduled timeslot - please contact the instructor.
• The lecture notes for 2005 are now available.
• Students who are not regularily enrolled can apply for the course by filling out an application form -- please bring this form with you to class - so that I can expedite its processing (since normally this application should be submitted in advance of the course.
• Note shifted afternoon hours on thursday - to avoid conflict with 2G1305
• 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"

2G1325/2G5564 Practical Voice Over IP (VoIP): SIP and related protocols (Röst över IP (VoIP) i praktiken: SIP och relaterade protokoll) is a 5 point course designed for advanced undergraduates (2G1325) and graduate (2G5564) students; especially those in the Telecommunication Graduate Program or the International Masters Wireless program.

Advanced undergraduates should have completed the course 2G1305 (Internetworking) or 2G1701 (Advanced Internetworking) or an equivalent course with a grade of 4 or 5 and obtain permission of the instructor.

Information is available on:

• Aim
• Prerequisites
• Contents
• Schedule
• Literature and Course Material (Textbook, Reference books and other references)
• Lecture Plan and Lecture Material (OH slides)
• Examination Requirements and Registrations
• Staff Associated with the Course
• Registering for the Course
• Other on-line Course Material (More References)
• Announcements
• Previous versions of the course

Aim

This course will give both practical and general knowledge concerning Voice over IP. The emphasis will be on the underlying protocols. After this course you should have some knowledge of these protocols: what they are, how they can be used, and how they can be extended. You should be able to read the current literature at the level of conference papers in this area.

As with the Internetworking course 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 you 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.

You should be able to write papers suitable for submission to Globecomm, Voice on the Net (VON), and other conferences and journals in the area. This course should prepare you for starting an exjobb in this area (for undergraduate students) or beginning a thesis or dissertation (for graduate students).

Prerequisites

• Telesys, gk or Datorkommunikation och datornät/Data and Computer Communications or equivalent knowledge in Computer Communications; Internetworking; and permission of the instructor

Students considering participating in this course should contact the instructor.

Contents

This course will focus on the protocls 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.

The course consists of 10 hours of lectures and an assigned paper requiring roughly 50h of work by each student.

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 50h of work by each student (5 p)
• Registration: 9-May 2005, to maguire@it.kth.se with the subject: 2G1325 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 "U".
  • 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 24 May 2005 + oral presentations scheduled from 8:00-17:00 on 3-June 2005 in room 437
  • Send email with URL link to maguire@it.kth.se
  • Late assignments will not be accepted
  • Note that it is pemissible 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 for 20 minutes, followed by 10 minutes of discussion. You only need to attend the day you present.

Grades: U, 3, 4, 5

Literature

Main Text-Book

The course will mainly be based on the book: Luan Dang, Cullen Jennings, and David Kelly, Practical VoIP: Using VOCAL, O'Reilly, 2002, ISBN 0-596-00078-2.

The second book is: 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

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)

Supplementary readings

• John Alexander (Editor), Chris Pearce, Anne Smith, Delon Whetten, Cisco CallManager Fundamentals: A Cisco AVVID Solution Cisco Press, 2001, ISBN: 1-58705-008-0.
• Gonzalo Camarillo and Jonathan Rosenberg, SIP Demystified McGraw-Hill Professional Publishing, 2001, ISBN: 0-07-137340-3.
• Daniel Collins, Carrier Grade Voice Over IP McGraw-Hill Professional Publishing, 2000, ISBN: 0-07-136326-2.
• Jonathan Davidson, James Peters, Brian Gracely (Contributor), Jim Peters, Voice over IP Fundamentals, Cisco Press, 2000, ISBN: 1-5787-0168-6.
• Jonathan Davidson (Editor), Tina Fox (Editor), Phil Bailey (Editor)ConCon Deploying Cisco Voice Over IP Solutions, Cisco Press, 2001, ISBN: 1-58705-030-7.
• Bill Douskalis, Putting VoIP to Work: Softswitch Network Design and Testing, Prentice Hall, 2002, ISBN 0-13-040959-6.
• Bill Douskalis, IP Telephony: The Integration of Robust VoIP Services, 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, 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, Cisco Press, 2001, ISBN: 1-58705-050-1.
• Mark A. Miller, Voice over IP Technologies: Building the Converged Network, Hungry Minds, Inc., 2002, ISBN 0764549073.
• Daniel Minoli, Delivering Voice over IP Networks, 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 http://www.upgrade-cepis.org 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
• William C. Hardy, "VoIP Service Quality: Measuring and Evaluating Packet-Switched Voice", McGraw-Hill, January 2003, 317 pages, ISBN: 0071410767. (note the reviews are very mixed on this book)
• Paul Mahler, VoIP Telephony with Asterisk, Signate, San Francisco, CA, 2004. ISBN 0-9759992-0-6

Useful URLs

• J. Loughney and G. Camarillo, Authentication, Authorization, and Accounting Requirements for the Session Initiation Protocol (SIP), RFC 3702, February 2004
• J. Rosenberg, A Session Initiation Protocol (SIP) Event Package for Registrations, RFC 3680, March 2004
• P. Faltstrom and M. Mealling, "The E.164 to Uniform Resource Identifiers (URI) Dynamic Delegation Discovery System (DDDS) Application (ENUM)", RFC 3761, April 2004.
• J. Peterson, "enumservice registration for Session Initiation Protocol (SIP) Addresses-of-Record", RFC 3764, April 2004
• O. Levin, "Telephone Number Mapping (ENUM) Service Registration for H.323", RFC 3762, April 2004
• vovida.org 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 of San Jose State University is teaching a course "EE284 Convergent Voice and Data Network" during Fall 2002 that also use this same book.
• Henning Schulzrinne's Session Initiation Protocol (SIP) web pages
• IETF SIP Working group
• IP Telephony
• SIP Forum
• SIP Center
• SIP Products at Pulver.com
• VoiceTronix analog line cards
• Voxilla.org hosts a collection of pointers to various open source telecom software projects for use with the GNU/Linux operating system
• GNUComm pre-release versions of some GNUComm Components:
  • GNU Bayonne, - 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
• Some SIP related Student Projects done under the supervision of Prof. Henning Schulzrinne
• Columbia InterNet Extensible Multimedia Architecture CINEMA
• NIST-SIP 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. See also http://www.research.att.com/info/Projects/mmdump
• 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
• End-to-End delay: http://wwwtvs.et.tudelft.nl/people/piet/papers/e2edelayripe_IEEE.pdf see also http://www.fokus.gmd.de/research/cc/glone/projects/cost263/meetings/09-namur/techdocs/Van-Mieghem-slides.pdf
• PIMRC paper on VoIP over Mobile IP
• Grandstream NetworksSIP phones and analog telephone adpators
• SIPphonea SIP service operator

Schedule

The schedule for lectures for 2G1325/2G5564 Practical Voice Over IP (VoIP) are shown below (Note that in the following "xx" means "xx:00", not "xx:15".):

Note that Sal E is in the Forum building in Kista (description and map).

Lecture Plan and Lecture Material (OH slides)

Note that the lectures will occur in a very intensive fashion to accommodate graduate students coming from elsewhere in Sweden.

version of lectures for 2005(2,146kB)

Staff Associated with the Course

• Lecturer (kursansvarig, föreläsare): Prof. Gerald Q. Maguire Jr. (maguire@it.kth.se)
• Administrative Assistant -- for administrative questions: recording of grades, ... contact - to be annouced

Registering

Use the normal process for registering. For most students this means you should speak with your study advisor (studievägledare.

Previous versions of the course

• 2004

Other on-line Course 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

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.

Sources for Further Information

• 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

Some ideas to investigate

• Lars Aronsson <lars@aronsson.se> in e-mail to the elektrosmog mailing list on 25 October 2002, ask if it would be possible to have a utility which would run in the background and display roundtrip time and jitter. He suggests displaying a "... `signal quality" meter on the screen, a simple dial from 0-3 (red, useless for VoIP), 4-7 (yellow, OK) to 8-10 (green, excellent)."
  Consider how you might use the information from the RTCP traffic to provide such feedback. Could you provide this via XML to a Cisco 7960 phone in real-time? Could you provide it at the end of a call, just as there is an application which queries the user after a call about the quality and uses this to compute a MOS rating.
  (For more info.)

Page History

© Copyright 2004, 2005 G.Q.Maguire Jr. (maguire@it.kth.se)
All Rights Reserved.
Last modified: 2005-06-01 20:31:28 MET DST 2005