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2007

VT 2007, 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: Thu Apr 21 17:57:28 CEST 2011

Announcements

  • The presentation should not be more than 15 minutes, given the 20 minute timeslot this gives time for a couple questions and changing presenters.
    I'm assigning timeslots as I correct papers.
  • All the papers which have been received has been assigned time slots for presentation in Aulan as per the following schedule on Friday 25th May
    8:00-8:20 empty
    8:20-8:40 Bridging SIP and Skype
    8:40-9:00 Adding context to SIP: CPL vs. B2BUA
    9:00-9:20 SIP Security Analysis
    9:20-9:40 Eyes on VoIP: A Description of the Centralized and the P2P Approach(es)
    9:40-10:00 Using SIP for both mobility and QoS
    10:00-10:20 SIP over WiFi
    10:20-10:40 The application of RTSP in Streaming Media
    10:40-11:00 NAT traversal in SIP
    11:00-11:20 The Use of IMS in Merging VoIP an 3G
    11:20-11:40 Voice over IP Security
    11:40-12:00 Conversion of Email into Instant Message and deliver to last hop
    12:00-12:20 Security Issues for VoIP
    12:20-12:40 VoIP over GPRS
    12:40-13:00 P2P audioconferencing using SIP (slot 2)
    13:00-13:20 Integration of VoIP with Mobile Service
    13:20-13:40 A SIP-based presentation support application
    13:40-14:00 Translators and Mixers
    14:00-14:20 Home Remote Control system on Applicance based on SIP
    14:20-14:40 Telephony over IP
    14:40-15:00 (late) Security consideration[s] for Multimedia Streaming over VoIP
    15:00-15:20 Peer-to-peer (P2P) SIP
    15:20-15:40 A Study on ENUM
    15:40-16:00 (late) The Future of Voice over IP as a Business
    16:00-16:20 Voice over IP (VoIP): The Billing perspective
    16:20-16:40 Security Features in SIP
    16:40-17:00 Mobile Alert Monitoring via IMS
    17:00-17:20 Friendly FON: Possibilities of Making a SIP UA on a Gumstix Embedded Computer
    17:20-17:40 Traversal of SIP message through NAT
    17:40-18:00 (late) Why SIP: a discussion about SIP, Skype, and Mobile Telephony
    18:00-18:20 SIP, SKYPE OCH KORSNIN UTAV NAT på 60 minuter
  • Course schedule conflict with 2G1723: GSM Network and Services on Thursday March 29. I will end the lecture on the afternoon of 29 March at 15:00 (rather than 16:00), so that students can go to the 15:00-17:00 2G1723.c Lab session. I hope that this will resolve the problem. -- It ended up that 2G1723 moved its lab.
  • For students who are looking for examples of papers - see the ACM Sigcomm 2005 proceedings - which are in the Computer Communication Review, Volume 35, Number 4, October 2005.
  • 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.
  • 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 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.

Learning Outcomes

Following this course a student should be able to:

  • Understand the relevant protocols (particularily 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 intecept)
  • 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 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; or 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: Monday 9-April 2007 at 23:59, 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 Friday 11-May-07 at 23:59 + oral presentations scheduled Friday 25-May-07 from 08:00-18:00 in Aula.
    • 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

"komplettering" - students who do not pass can submit a revised version of their paper (or a completely new paper) - which will be evaluated.

Code of Honor and Regulations

KTH has a common code of honor and regulations (see Code of Honor and Regulations).

Literature

Main Text-Book

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, 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)

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".):

DateTimeRoomNotes
Thursday 29-Mar-07 10:00-12:00 Aula Föreläsning 1
Thursday 29-Mar-07 13:00-16:00 Sal E Föreläsning 2
Friday 30-Mar-07 10:00-12:00 Sal D Föreläsning 3
Friday 30-Mar-07 13:00-16:00 Aula Föreläsning 4
Friday 30-Mar-07 15:00 from Efftel will speak about their VoIP solution as an example.

Note that Aula, Sal D, and Sal E are 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 2007(~2.2MB)

Staff Associated with the Course

Registering

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

Other on-line Course Material

Gizmo Project, SIPphone, Inc.

Google Talk voice-chat

PeerMe, PeerMe, Inc.

Yahoo! builds upon Dialpad acquisition to offer VoIP via its messanger

MCI Web Calling for Windows Live Call

Stefano Ventura, VoIP&Security for Enterprise, 8.11.2005 - a very nice introduction to VoIP security (in french)

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.

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.

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.

Sources for Further Information

Previous versions of the course

Page History

2007.05.22 added schedule for oral presentations
2007.03.27 added lecture notes for 2007
2007.03.27 correct due dates for 2007
2007.03.05 changed first afternoon session to Sal E
2006.12.13 added pointer to Sipp and sipsak
2006.12.06 added voip-4d link
2006.11.07 added Per's presentation
2006.11.05 added course dates and rooms for 2007
2006.08.03 added information about XCON and CONFIANCE
2006.06.13 First version for 2007

© Copyright 2004, 2005, 2006, 2007 G.Q.Maguire Jr. (maguire@it.kth.se)
All Rights Reserved.
Last modified: Thu Apr 21 17:57:28 CEST 2011

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