2004

VT 2004, 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: 2005-02-08 19:52:04 MET 2005


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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: 10-May 2004, 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.
    • The report may be in the form of a collections of paper, 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 2004 + oral presentations scheduled from 31-May 2004 to 04-June 2004
    • 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

All lectures and the oral presentations will take place in Sal E (formerly known as Sal 5) - this room is located on the 5th floor of the Forum building in Kista.

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".):

DateTimes
Thursday 01-Apr-04 10:00-12:00 and 13:00-16:00
Friday 02-Apr-04 10:00-12:00 and 13:00-16:00

Final oral presentations will be made for both this course and another course (2G1330: Mobile and Wireless Network Architecture) on the following days:

Mon 31-May-04	8:00-17:00 2G1325 and 2G1330
Tue 01-Jun-04	8:00-17:00 2G1325 and 2G1330
Wed 02-Jun-04	8:00-17:00 2G1325 and 2G1330
Thu 03-Jun-04	8:00-17:00 2G1325 and 2G1330
Fri 04-Jun-04	8:00-17:00 2G1325 and 2G1330

The split between the courses will be based on the numbers of students in each course who have submitted reports.


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 2004


Staff Associated with the Course


Registering

to be added


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.

An example of a written report submitted in 2004:
Andreas Ångström and Johan Sverin, VoiceXML, it appears here with permission of the authors.

Another example of a written report submitted in 2004:
A real-time tool to display the quality of voice communications over IEEE 802.11b networks, it appears here with permission of the author.


Sources for Further Information

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.

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

DateUpdate
2005.02.08 removed admin contact
2004.06.30 Added another example paper
2004.06.10 Added example paper
2004.03.30 Added version of lecture notes for 2004
2004.03.23 Noted that Sal 5 is now known as Sal E.
2004.02.27 First version for 2004

© Copyright 2004, 2005 G.Q.Maguire Jr. (maguire@it.kth.se)
All Rights Reserved.
Last modified: 2005-02-08 19:52:04 MET 2005

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