SIP Tutorial

Presenters:
Stephen Kingham
Stephen.Kingham@aarnet.edu.au
And
Prof Dr Quincy Wu (aka Aaron Solomon)
solomon@ipv6.club.tw
 VoIP Basics
Presenters:
Stephen Kingham
Stephen.Kingham@aarnet.edu.au
 Copyright Stephen.Kingham@aarnet.edu.au 2006

This work is the intellectual property of the author.

Permission is granted for this material to be shared
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this copyright statement appears on the reproduced
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republish requires written permission from the author.

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 Outline
Introduction
What is VoIP
Round table introductions
10:30 Morning tea
11:00 SIP Protocol, some demonstrations
12:30 Lunch, 90 minutes
14:00 SIP Protocol
15:30 Afternoon Tea
16:00 Some case studies and questions
17:30 or earlier FINISH
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Stephen Kingham@aarnet.edu.au
 Other relevant talks at APAN Tokyo 2006
Monday 23 Jan
SIP User Agents Configuration and Fault Finding
Speaker: Quincy Wu
SER Configuration and SIP Peering including ENUM
Speaker: Stephen Kingham
From Taiwan SIP Mobility in IPV4/IPV6 Network
Speaker:
Using Radius and LDAP with SER SIP Proxy for user Authentication
Speaker: Nimal Ratnayake
9:30 Wednesday 25 Jan
Global SIP Dialling Plans (Ben Teitelbaum and Dennis Barron)
16:00 Wednesday 25 Jan
APAN SIP-H.323 Working Group BoF
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Stephen Kingham@aarnet.edu.au
 What is
IP Telephony, VoIP and VIDEO?

Presenter: Stephen Kingham

Stephen.Kingham@aarnet.edu.au
 Outline
Realise there is a difference between:
VoIP
IP Telephones PABX
IP Telephones roaming
Video

In terms of
Design
Support
View to the user
Business Case
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Stephen Kingham@aarnet.edu.au
 VoIP Standards
1. In 1995 we got the standard H.323. This is a Video
Standard from the Carrier world and is based on
ISDN.
2. In June 2002 we got SIP from the Internet Standards
body (IETF). It uses all the other Internet standards.
Is Video, Presence, and Instant Messaging, plus
more. Is extreamly simple (read scary with
potential).
3. And we have some proprietary protocols/technology
(read painful).

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Stephen Kingham@aarnet.edu.au
 Telephones BEFORE the 2000s
Basic Telephone service

PABXs generally provided by Carriers, usually on Carrier

recommended PABX equipment.

In Universities it was provided by the Buildings and Grounds

departments in Universities.

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Stephen Kingham@aarnet.edu.au
 Telephones in the 80s - deregulation
Still Basic Telephone service
Shared structured cabling between LAN and Telephones

Generally still provided by Carriers. Some private networks using

TDM and some tie-lines and voice compression.
More choice of PABX platform.

(Tele)Communications Section created by bringing the Voice and

Data Communications together as separate Sections under one
management group.

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Stephen Kingham@aarnet.edu.au
 Telephones in 2000-2004 H.323 and VoIP
Still Basic Telephone service
But VoIP used to link PABXs together, and
some VIDEO conferencing.

Replaced TDM based.

Huge improvement in reliability.

VoIP needs WAN Section to work with Voice Section.

VoIP is NOT IP Telephony

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Stephen Kingham@aarnet.edu.au
 VoIP is like the Wide Area Network
Technically VoIP contains the
Routeing
Servers, such as Voice Mail, IVR etc
Billing
QoS on WAN
Support involves supporting Level 2/3
and Carrier connections (not Users!)
Business case is around
Toll By Pass
Supporting IP Telephones and or Video

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Stephen Kingham@aarnet.edu.au
 VoIP

SIP & H323

PABX Voice AARNet SIP &Voice
H323 PSTN
GATEWAY Internet with Voice
GATEWAY PABX
Carrier
QoS bandwidth GATEWAY

SIP Server
or H.323 Gatekeeper Other
Translate telephone numbers to IP addresses
advanced IP
network

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Stephen Kingham@aarnet.edu.au
 2000-2004 here comes H.323 and
Proprietary protocols for IP Telephones
Proprietary IP Telephones deployments:
H.323 too hard (although Avaya did it).
whole University Campuses (some of the largest Universities in Australia).
Some hybrids (IP Telephones with PABX left) and some entirely IP
Telephony.
IP Telephony based on top of solid VoIP network.
Long term better investment and large reductions in adds moves and
changes

VoIP needs WAN Section to work with Voice Section.

IP Telephony needs LAN Section to work with Voice Section
There is a difference between VoIP and IP Telephony
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Stephen Kingham@aarnet.edu.au
 IP Telephones are like LOCAL Area Network
Technically it contains the
PABX replacement
Security
IP Phones
Power to IP Telephones
Billing
QoS on LAN
Access to emergency services
Support involves supporting Users
Business case is around
PABX Replacement
Reduce Costs for Adds Moves and Changes
Improved productivity and integration
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Stephen Kingham@aarnet.edu.au
 IP Phone

SIP & H323

PABX Voice AARNet
GATEWAY Internet with
QoS bandwidth

SIP Server
or H.323 Gatekeeper
Translate telephone numbers to IP addresses

SIP UA
(IP Telephone)

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Stephen Kingham@aarnet.edu.au
 PABX IP Telephones : Emergency Services
Make sure calls to Emergency Services (eg 119 in Japan, 911 in USA, 000 in
Australia, etc) go to the VoIP Gateway that is at the same site as the IP
Telephone.

Calls to 000
Site A Site B

Call Manager
Server

NO

PSTN Voice Voice PSTN

Carrier GATEWAY GATEWAY Carrier

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Stephen Kingham@aarnet.edu.au
 Telephones in 2005+ The impact of SIP and
3rd party Carriers - The revolution begins!
Explosion of SIP UAs and PABXs into the market.
Many 3rd party providers of sip: accounts.
Some proprietary solutions (eg Skype) plus some who lock customer in using
SIP (eg MSN and Yahoo) sometimes called islands.
All the IP Phone and traditional PABX vendors are moving to SIP.
SIP based PBXs with exceptional capabilities and features, at a fraction of
traditional TDM switches.
Control given back to the user.

Introduction of the Unix System Administrator (and programmer) skills into the
Voice Section.
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Stephen Kingham@aarnet.edu.au
So in summary we have described three characteristics:
VoIP
WAN, Gateways, QoS, MCUs, Toll Bypass, different support processes.
IP Telephones
LAN, PABX stuff, Emergency Services, built on VoIP, different Business
Case to VoIP, different support processes.
Roaming IP Telephone
A different type of IP Telephone!
Issues to be determined.

And lets not forget that V stands for Video, Instant Messaging and
Presence as well as Voice, plus who knows what else
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Stephen Kingham@aarnet.edu.au
 Affordable SIP products (NOT H.323)
Basic SIP IP phones below US$75
802.11 phones (need certificate support)
Video phones
Speakerphones
PDAs with SIP software
MAC, Unix, and MSoft.

Combination of Stephen Kingham and Quincy Wus talk, www.apan.net Cairns 2004
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 Also SIP Clients
PDAs with SIP software
MAC, Unix, and MSoft.

Combination of Stephen Kingham and Quincy Wus talk, www.apan.net Cairns 2004
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Stephen Kingham@aarnet.edu.au
 SIP based PABXs (The SIP Server)
SIP is so easy to develop in.
Many quality Open Source SIP PABXs.
Some of the VoIP Carriers use these Open Source Products!
They include Call Routing, Forwarding, IVR, and Voice Mail.
All the PABX Vendors are moving to SIP based technology.
All the Carriers are deploying their VoIP and IP Telephone Services
using SIP Technology.
With SIP it is easy to mix and match products.
SIP is really easy to support.

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Stephen Kingham@aarnet.edu.au
 Here is a possible view of the future (today commercial product)
a full Voicemail System in 20 lines of Perl (Slipper HelperApp::)

#!/usr/bin/perl -w
use strict;
use Slipper::HelperApp;
my $stream = Slipper::HelperApp -> new_stream (shift, shift);
if (! ref $stream) {
print $stream . "\n";
exit 0; Slipper is an example of a modern
} commercial PABX Call Server up to even
my $return = $stream -> find_vm_target;
for a small Carrier
if ($return !~ /^200/) {
print $return;
exit 0;
}
$stream -> report_port;
$stream -> play_audio ($stream -> {'VM Greeting'});
$stream -> play_audio ('vm/pling.au');
my ($dtmf, $message) = $stream -> record_audio;
exit 0 if (! defined $message);
$stream -> send_vm ($message);
23 exit 0;
Andrew.Rutherford@iagu.net
 The impact of SIP : SIP based PBXs
Some of these offer exceptional features and capacities

SIP Express Router (SER) Open Source from http://www.openser.org/

(was www.iptel.org).
one config file and mysql
SIPx (Open Source)
Asterisk is not really SIP or H.323
does some nasty things to the codec negotiations
but it is very popular.
Supports Gateway cards to PSTN, H323-SIP GW, IVR, and Voice Mail.
Many config files.
Yate (Yet Another Telephone Engine) http://yate.null.ro/pmwiki/
Does many things and claims to have a great H.323-SIP gateway.

There is the start of an explosion of very good quality SIP PBXs.

24 Stephen Kingham@aarnet.edu.au
 All the Vendors moving to SIP
NEC
Avaya
Cisco new Call Manager is SIP in the core not skinny.
Nortel
Microsoft (PABX functionality soon)
An Australian Product called Slipper by IAGU.

Both Avaya and Cisco integrate the PC with the IP Telephone

to make a user friendly Video phone.

With SIP it is easy to inter-work.

Voice mail and IVRs are very easy.

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Stephen Kingham@aarnet.edu.au
The impact of SIP providers of sip: accounts
Provide sip accounts like hotmail provides email accounts.
Free World Dial (fwd) fwd.pulver.com
www.atp.org (in Australia)
And many many more, impossible to estimate the number

Providers of closed sip accounts (is this unproductive behaviour?):

MSN
Yahoo
Skype is NOT SIP and has serious implications for integration and securty and it shows
us what the user wants!

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Stephen Kingham@aarnet.edu.au
 SIP based VoIP Carriers (too many to list)
ENGIN G02Call StanaPhone
AAPT iConnectHere SunRocket
Internode/Agyle InPhonex TeleSIP
ATP Lingo TeIIAX
TerraCall
Mutualphone
ITouchTone USA Datanet
MyPhoneCompany VoiceGlo
AOL Net2Phone VoicePlus
AT&T Nikotel VoiceWing (Verizon)
BroadVoice NuFone VoipJet
Broadvox Direct Packet8 Vonage
Dialpad QuantumVoice VoxFlow
Galaxy Voice SimpleTelecom WebPhone
Global Village SIPphone Yahoo
G02Call SIPphone ZipGlobal
Skype (not SIP) IIC (old ozemail)

There are some key questions to ask.

27 Source: Wil.Daniels@acu.edu.au, Steering Committee Member for the AARNet IPTEL Working Group
 VoIP Carriers that provide SIP: accounts
Provide OPEN sip accounts like hotmail provides email accounts.
Free World Dial (fwd) fwd.pulver.com
www.atp.org (in Australia)
www.iptel.org (home of Open Source SIP Server SER)
And many more

Providers of CLOSED sip accounts:

MSN
Skype (not SIP)
Most do not permit calls to or from other VoIP provides.

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Stephen Kingham@aarnet.edu.au
 SIP based VoIP Carriers
ENGIN, buy a black box from Dick Smith (no QoS).
AU10c (untimed) to any Australian number, AU29c/min to mobiles, free to another engin
user, AU3.5c/min to key international destinations.
Internode combined with the Internode ADSL (has QoS).
AU18c (untimed) to any Australian number, 30c/min to mobiles, free to another internode
number, AU15c/min to key international destinations.
Free World Dial (no QoS), provides a SIP account
Call other SIP addresses.
Call other VoIP Networks using an access code.
AARNet (with QoS)
AU6c plus AU1c per minute to 90% of Australians, 25c/min to Mobiles.
Only available to AARNet Member Organisations.
Standard telephone rates
around AU25c per local call,
Around AU X per minute for Long Distance.

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Stephen Kingham@aarnet.edu.au
An example: SKYPE is an island, and it is not SIP!
Has one good lesson: It shows what we need to do for the users!
Lots of negatives:
Proprietary (secret) protocol.
Major security accident waiting to happen as soon as someone reverse
engineers the protocol (ref www.voipsa.org/VOIPSEC - VoIP Security).
User has no control over their bandwidth, eg if they become a Skype Super
Node, other people will use your bandwidth.
Loose corporate identity, replaced with a skype identity.
Can not integrate with existing infrastructure such as PABX, Video conferencing,
Voice Mail, Room based Video, etc.
It is an island. To call out/in of the island you have to pay $money.

0.017 (about A$0.027c) per minute to Australian Numbers is more expensive than
AARNet and Engin.
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Stephen Kingham@aarnet.edu.au
 SIP will impact desk top Collaboration
Two problems seam to dog video conferencing, getting through firewalls and routing

SIP & H 323 SIP b2bua, and a

PABX Voice University
H.323 GK in Proxy
GATEWAY Network
mode

H.323 Room
based Video AARNet
SIP & H.323 Internet with
MCU QoS bandwidth
Desktop Video

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Stephen Kingham@aarnet.edu.au
 Other Security issues
Generally all UDP hi ports need to be opened.
Alternative is to use a b2bua that has visibility to the inside and
outside. Or a b2bua can be used to solve NATing together with
an encrypted tunnel.
Another solution are statefull firewalls, and they are slowly
improving. Ask if your firewall supports SIP and if it also supports
QoS.
Encrypted tunnels is another viable solution.

Always be mindful you are working with delay and jitter sensitive
communications.
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Stephen Kingham@aarnet.edu.au
 What the customer wants?

Could Universities start loosing

their customers
to 3rd party providers?

Has this already started?

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Stephen Kingham@aarnet.edu.au
 SIP FORKING (native to SIP)
Never need to forward phones to other phones again!!!!
This is a big mindset change for the user.
Someone calls 02 6222 3575
Voice PSTN
INVITE 1 GATEWAY Carrier
Voice SIP Proxy 1747
PSTN Server 614194
GATEWAY
Carrier INVITE
St Extn 3
ep 575
he
PABX
Delay
INVIT

n.K
ing INV
ha ITE
ed

m@
E

aa
rn
et.
ed
u.a
Voice u
Mail
34 SIP UA
Stephen Kingham@aarnet.edu.au
 The Revolution has started
Control given back to the user. No more forwarding calls.
Presence and instant Messaging.
Introduction of the Unix System Administrator (and programmer) skills
into the Voice Section.
Lots of hype and confusion in the market place. watch out for
destructive events (skype, and SPIT).
The telephone will be unrecognisable.

Look forward to lots of sipping

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