Tutorial on SIP and SIP SER server

Last update:19 January 2005. This page is under constant development.

Back to the AARNet/Grangenet Workshop Registration page.
Back to APAN web site

This page will provide information and details required by prospective participants for a hands-on SIP workshop/tutorial. Details will include:

Objectives
Preparation needed by Delegates and what is being supplied at the Tutorial
Course outline
Details of when and where the workshop will be held.
Technical description of VLANs, LAN and WAN provided at the Tutorial
Links to other resources.

Any questions or suggestions should be directed to Stephen.Kingham@aarnet.edu.au

WHO SHOULD ATTEND:

Managers so they can make informed strategic decisions for Telephony, PABX replacement, Video over IP, and new services that will be enabled by SIP: Presence and Instant Messaging.
Video, PABX and Telecommunications Engineers and Technicians to get a fast start on SIP.
Non-AARNet Members are allowed to attend but unreserved preference will be given to AARNet Members.

OBJECTIVES:

To appreciate that SIP is the replacement for existing H.323 and proprietary protocols for IP Telephones, VoIP, and VIDEOoIP.
Raise awareness of new user services that will be enabled by SIP: Presence and Instant Messaging.
Teach the SIP protocol in detail.
Familiarise with various SIP devices and Servers, such as Telephones, Video, Gateways to PSTN/ISDN, Voice Mail, IVRs, and Conferencing.
Hands-on component to configure SIP Software on PCs, IP Telephones, and Servers.
OPTIONAL: Participants can bring, or have access to via the Internet, a Linux/FreeBSD PC with the necessary software packages installed to support them making VoIP calls between participants in the workshop and externally. See Preparation below

PREPARATION NEEDED BY DELEGATES

No preparation is required.
Delegates are encouraged to bring a notebook to try out some of the SIP Software and tools to fault find SIP. A notebook will be necessary to log onto any of the provided SIP Servers.
Optional:
- Delegates are encouraged to have their own Unix server built with the SER SIP Proxy Server installed before the Tutorial/Workshop. There will be approximately 3 Unix servers for delegates to use during the Tutorial for those who do not have their own.
- A template on building a FreeBSD or a RedHat Fedora Core 1 Unix server are available here. This information includes what security changes you need to do to allow access from the workshop back to a Unix server you leave connected to you own network
- A template to have SER installed is available here.
- If you have your own preferred Unix Version or alternative SIP Server such as Asterisk Server then you are welcome to use those.

WHAT IS BEING SUPPLIED:

There will be a range of SIP Telephones available for Groups to use.
We will have about 3 servers for Delegates who do not bring their own.
Network access to AARNet/GrangeNet.

COURSE FORMAT AND APPROACH:

See outline for more detail.

The Delegates will be split up into groups of between 2 and 4 depending on the numbers, we will keep Delegates from the same institution together unless specifically requested not to.
The Workshop will be informal and flexible, changing to the capability of the Delegates.
The tables will be arranged in a horse shoe format (if possible).
Two projector screen will be used. One will be for the formal slides and the other used to demonstrate.
Hands on:
- Each group will have a working SIP Telephone and at least one IP phone for them to configure.
- Each group will be able to install a public domain SIP client.
- Groups can use their own SIP Servers to configure.
- The provided SIP Servers will be built and configured leaving delegates to add users and experiment with routing.

COURSE OUTLINE:

This outline will change.

Start Duration	Title Objective	Speaker
9:00 10minutes	Introduction	srk
9:10 20 minutes	Review VoIP, IP Telephony and Video In terms of where the technology can be applied (WAN and/or LAN), the sorts of applications (toll bypass or PABX replacement) will have entirely different issues in terms of design, QoS, Support, Implementation and Business Cases.
9:30 40 minutes	Introduction to SIP (slides, call flows by Zultys, MSWord doc) - basic introduction through to detailed examination of the protocol. - examination of calls made at the workshop. Largely a formal presentation with some demonstration. Some of the SIP detail will be covered in other sessions that follow.	srk
10:10 15 minutes	Morning Tea
10:30 40 minutes	Installation and basic configuration of SER Proxy Server (slides, link to install guide) - Overview of installation of SER - Configuration of SER to make and receive calls from the PSTN and SIP User Agents (UAs). Presentation, demonstration and hands-on.	srk and reh
11:10 40 minutes	Installation and basic configuration of Asterisk Proxy Server - Overview of installation of Asterisk - Configuration of SER to make and receive calls from the PSTN and SIP User Agents (UAs). Presentation.	Dependent on availability of a speaker
11:50 30 minutes	Installation and configuration of various SIP Clients, Gateways and Servers (Link to various configuration templates for User Agents) Between three and four UAs will be reviewed. Cisco VoIP Gateways Zultys telephones Cisco 7960 telephones Polycom telephones Windows Messenger 5.0 Radvision viaIP400 MCU SIPC France telecom eConf NBEN from TWAREN - demonstration only. X-lite Delegates will have registered various User Agents (SIP Clients) to their SIP Proxy server and be able to make and receive calls.	srk and reh
12:20 40 minutes	Lunch
12:40 20 minutes	Some more SIP features not covered yet (slides) Understand Authentication in SIP & Authorisation in SER. Instant Messaging Presence	srk
13:00 40 minutes	Operating and Fault Finding SIP (slides) The following problems will be reviewed and some demonstrated: User cannot register User cannot make a call User cannot receive calls Call connects but no Audio or Video in one or both directions The following tools will be demonstrated: Where to find and use the SER logs and use secret ngrep to view traffic to from Server How to use Ethereal to decode SIP myphpadmin for managing the mysql database	srk
13:40 30 minutes	Peering SIP Networks (slides) Review: Dialing plans using ITU-T dialing Dialing Plan using SIP URI Delegates will configure their SER server to be part of the SIP.edu project using an external module or use a populated mysql database on the SER server Introduce and describe how ENUM could be used Introduce Telephony Routing over IP (TRIP) (RFC 3219) Areas for future work and collaboration in APAN and Internet2	srk
14:00 160 minutes	EXERCISES During this session delegates get to work on a variety of exercises and key issues can be reviewed as a group. Individuals or groups will be left to go through selected exercises with assistance provided by presenters. Towards the end the presenters will go through each exercise. Exercise 1: Configure a new user on SER able to make calls to mobile telephones Access to Server Username and password chosen by delegate, hint "secret add" and "secret acl". Two working phones, at least on the server to test Exercise 2: Configure a second line on a working IP Phone to register to a SIP Server Access to configure phone A valid username and password A working IP Phone with at least one working line Exercise 3: Install Windows Messenger as a SIP UA Delegate must provide PC A working SIP Server Access to a tftp boot server or configure manually Valid username password Exercise 4: Install a Cisco 7960 phone A Cisco 7960 A working SIP Server Access to a tftp boot server or configure manually Valid username password Exercise 5: Install a Zulty 2x2 or 4x4 phone A Zulty 2x2 and or Zulty 4x4 A working SIP Server Access to a tftp boot server or configure manually Valid username password Exercise 6: Install a Polycom IP 500 phone A Polycom IP 500 A working SIP Server Access to a tftp boot server or configure manually Valid username password Exercise 7: Install a Avaya 4602 phone An Avaya 4602 A working SIP Server Access to a tftp boot server or configure manually Valid username password Exercise 8: Configure a SER SIP Server to route a specific number range to another SIP Server Access to one SIP Server Address of a second SIP Server Working phone on each Number range to route Exercise 9: Configure a SER SIP Server to operate as sip.EDU, ie make a telephone ring using a sip uri. Access to one SIP Server with a DNS SRV record Working PSTN phone on SIP server Exercise 10: Configure a SER SIP Server to "fork" an incoming call to multiple destinations, destinations could include a SIP UA on same Server, one PSTN phone, and a UA on another SIP Server.. Access to one SIP Server with a DNS SRV record Working PSTN phone on SIP server SIP Server can make calls to PSTN
14:30	Afternoon Tea
	Continuation of Exercises
16:35 40 minues	A Mixed bag of issues and Services (slides) Voice Mail Interactive Voice Response (IVR Units) Computer Telephone Integration Presence
17:15 15 minutes Finish at 17:30	Conclusion/wrap up Slides (ppt 550kBytes)	srk

PROBABLE SUBJECTS FOR ADVANCED SIP WORKSHOP:

Start Duration	Title Objective	Speaker
	Working with Network Address Translation How to get around NATs
	Integration with H.323 Review these options: H.323open
	IPV6 integration Slides (ppt XXXBytes, pdf XXXBytes)
	Radius Accounting

Templates for the equipment and the LAN at the workshop

The page on the following link has:

IP Addresses used by the Groups in the Workshop
SRV SIP entries for the each Group
PSTN telephone numbers allocated to the IP Phones at the Workshop and the Telephone routing. Groups use the above two to set up telephone routing from their Group's SI PServer to others.
Configuration files for each of the IP Telephones in the Workshop. Groups use these to quickly configure there IP Telephones on their server.
Workshop configuration templates for Workshop participants

Where the workshops have and will be held

4 Jul 2004 Cairns Australia: Workshop was first held at Questnet 2004 / APAN Cairns 2004. Presented by AARNet, NCTU, and Zulty's.
30 Oct 2004 Austin Texas USA: near identical workshop at Internet2 Members Meeting 2004. Presented by Internet2 VoIP working group, Internet2 VoIP ITEC and Texas A&M.
27 Oct 2004 Room 104 Building 4 UTS Broadway Sydney Australia 2004 October 2004. Presented by AARNet.
23 Nov 2004, Canberra Australia, AARNet offices CSIRO Building 9 Banks Street Yarrulumla Presented by AARNet.
1 Dec 2004, Melbourne Australia, Grand Hyatt, Organised by Terrapinn. Presented by AARNet.
3 DEC 2004, Melbourne Australia, ACU, Presented by AARNet.
25 Jan 2005 Bangkok Thailand at the APAN Bangkok 2005 Meeting. Presented by AARNet and NCTU.
10 February 2005, Brisbane Australia, Queensland University of Technology Australia, Presented by AARNet

Future

15 February, Flinders Univeristy Adelaide Australia, Presented by AARNet
18 February, Edith Cowan University Perth Australia, Presented by AARNet.