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)

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
• 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.

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

• 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 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

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

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.

• 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.

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

• How to use SER with CPL
• To use SER with TLS and minisip - change your TLS method in the openser.conf file to: "tls_method = SSLv23" (thanks to Pjothi's comments in the minisip users mailing list on Fri, 3 Feb 2006
• 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
• VoP Security Forum has a tool: SiVuS - The VoIP Vulnerability Scanner
• Blue Box Podcast #22: SIP Security at IETF (part 1), VoIP security news, comments and more, April 7, 2006
• University of Naples, in cooperation with Ericsson Nomadic Lab in Helsinki, have released a first implementation of an XCON-compliant conferencing platform. The server side is based on Asterisk and a modified version of its MeetMe application, while the client side is based on Minisip. The system uses the Binary Floor Control Protocol (BFCP). The project is called CONFIANCE (CONFerencing IMS-enabled Architecture for Next-generation Communication Experience).
  For additional details about the proctocols see the IETF Centralized Conferencing (XCON) working group.
• Alberto Escudero Pascual and Louise Berthilson, "VoIP-4D Primer- Building Voice Infrastructure in Developing Regions, Translators: Anas Tawileh (Arabic), Johan Bilien (French). Available in English, Arabic, French, and Spanish.
• SIP debugging and testing
  • Sipp (sipp.sourceforge.net) - conformance testing tool
  • SIP Swiss Army Knife (SIP SAK) - a useful command line tool for SIP development and administration

Previous versions of the course

• 2006
• 2005
• 2004

Page History

© 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