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
- 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
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
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".):
| Date | Times |
|---|
| 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
| Date | Update |
|---|
| 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
