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2005

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)

Last modified: 2005-06-01 20:31:28 MET DST 2005

Announcements

NOTE

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

starting Title
08:00 Benefits and Problems with Pluggable Codec's for VoIP
08:20 Jäfouml;relse av klienter och protokoll f¨r IP-telefoni
08:40 B2BUA
09:00 Push-to-talk over Cellular (PoC)
09:20 A comparison between the end-to-end security in VoWLAN and the security offered in GSM and UMTS
10:00 ENUM
10:20 Roving Habitats
13:00 Security Analysis for the SIP-based VoIP
13:20 SIP over "Fast-track" TLS for mobile VoIP
15:20 Intra Voice over IP Telecommunication System
15:40 System and method for identifying a participant during a conference call with the user of source identifiers
16:20 Voice capabilities using J2ME
  • 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:

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

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

Useful URLs

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".):

Date Time Room Notes
Thu 21-Apr-05 10:00-12:00 Sal E Föreläsning 1
Thu 21-Apr-05 14:00-17:00 Sal E Föreläsning 2
Fri 22-Apr-05 10:00-12:00 Sal E Föreläsning 3
Fri 22-Apr-05 13:00-16:00 Sal E Föreläsning 4

Note that Sal E is in the Forum building in Kista.

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

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.

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

 

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

2005.06.01 added list of talks
2005.05.25 added note regarding timeslots
2005.04.27 added link NENA's IP Capable PSAP document
2005.04.17 added link to lecture notes for 2005
2005.04.13 added link application form for those not regularily enrolled
2005.02.25 added link map to room location
2005.02.19 added link to VOMIT utility and note about EVE
2005.02.16 added hours and room number for final oral presentations
2005.01.24 shifted afternoon hours on thursday due to overlap with 2G1305
2005.01.17 added SER and SEMS links
2004.11.22 updated administrative contact
2004.10.21 added lecture dates
2004.10.08 added link to SJphone, minisip, and iptel.org's list of softphones
2004.09.24 added link to previous versions of the course
2004.09.18 added links Asterik and Brekeke
2004.09.17 added link to Texas A&M IETC for VoIP
2004.07.15 added link about accessing the KTH e-library
2004.07.15 added link to Mongolia at TU-Berlin
2004.07.12 added another book and a link to Raj Jain's list of references
2004.07.03 added another example report from 2004
2004.06.28 First version for 2004

© 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

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