Transcript PowerPoint Presentation - Next Generation Networking and

SIP Tutorial
Presenters:
Stephen Kingham
[email protected]
And
Prof Dr Quincy Wu (aka Aaron Solomon)
[email protected]
VoIP Basics
Presenters:
Stephen Kingham
[email protected]
Copyright [email protected] 2006
This work is the intellectual property of 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
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•
17:30 or earlier FINISH
©Stephen [email protected]
Other relevant talks at APAN Tokyo 2006
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Monday 23 Jan
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9:30 Wednesday 25 Jan
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Global SIP Dialling Plans (Ben Teitelbaum and Dennis Barron)
16:00 Wednesday 25 Jan
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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
APAN SIP-H.323 Working Group BoF
©Stephen [email protected]
What is
IP Telephony, VoIP and VIDEO?
Presenter: Stephen Kingham
[email protected]
Outline
• Realise there is a difference between:
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VoIP
IP Telephones PABX
IP Telephones roaming
Video
In terms of
– Design
– Support
– View to the user
– Business Case
©Stephen [email protected]
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 [email protected]
Telephones BEFORE the 2000s
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Basic Telephone service
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PABXs generally provided by Carriers, usually on Carrier
recommended PABX equipment.
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In Universities it was provided by the “Buildings and Grounds”
departments in Universities.
©Stephen [email protected]
Telephones in the 80s - deregulation
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Still Basic Telephone service
Shared structured cabling between LAN and Telephones
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Generally still provided by Carriers. Some private networks using
TDM and some tie-lines and voice compression.
More choice of PABX platform.
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(Tele)Communications Section created by bringing the Voice and
Data Communications together as separate Sections under one
management group.
©Stephen [email protected]
Telephones in 2000-2004 – H.323 and VoIP
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Still Basic Telephone service
But VoIP used to link PABXs together, and
some VIDEO conferencing.
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Replaced TDM based.
Huge improvement in reliability.
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VoIP needs WAN Section to work with Voice Section.
VoIP is NOT IP Telephony
©Stephen [email protected]
VoIP is like the Wide Area Network
• Technically VoIP contains the
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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 [email protected]
VoIP
PABX
SIP & H323
Voice
GATEWAY
SIP Server
or H.323 Gatekeeper
Translate telephone numbers to IP addresses
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©Stephen [email protected]
AARNet
Internet with
QoS bandwidth
SIP &Voice
H323
PSTN
PABX
Voice
GATEWAY
Carrier
GATEWAY
Other
advanced IP
network
2000-2004 – here comes H.323 and
Proprietary protocols for IP Telephones
•
Proprietary IP Telephones deployments:
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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
©Stephen [email protected]
IP Telephones are like LOCAL Area Network
• Technically it contains the
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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 [email protected]
IP Phone
PABX
SIP & H323
Voice
GATEWAY
SIP Server
or H.323 Gatekeeper
Translate telephone numbers to IP addresses
SIP UA
(IP Telephone)
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©Stephen [email protected]
AARNet
Internet with
QoS bandwidth
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.
Site A
Calls to 000
Site B
Call Manager
Server
NO
PSTN
Carrier
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Voice
GATEWAY
©Stephen [email protected]
Voice
GATEWAY
PSTN
Carrier
Telephones in 2005+ The impact of SIP and
3rd party Carriers - The revolution begins!
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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.
©Stephen [email protected]
So in summary we have described three characteristics:
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VoIP
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IP Telephones
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WAN, Gateways, QoS, MCUs, Toll Bypass, different support processes.
LAN, PABX stuff, Emergency Services, built on VoIP, different Business
Case to VoIP, different support processes.
Roaming IP Telephone
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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 [email protected]
Affordable SIP products (NOT H.323)
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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 Wu’s talk, www.apan.net Cairns 2004
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Also SIP Clients
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PDAs with SIP software
MAC, Unix, and MSoft.
Combination of Stephen Kingham and Quincy Wu’s talk, www.apan.net Cairns 2004
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©Stephen [email protected]
SIP based PABXs (The SIP Server)
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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.
©Stephen [email protected]
Here is a possible view of the future (today commercial product)
a full Voicemail System in 20 lines of Perl (Slipper HelperApp::)
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#!/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;
}
my $return = $stream -> find_vm_target;
if ($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);
exit 0;
©[email protected]
Slipper is an example of a modern
commercial PABX Call Server up to even
for a small Carrier
The impact of SIP : SIP based PBXs
Some of these offer exceptional features and capacities
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SIP Express Router (SER) Open Source from http://www.openser.org/
(was www.iptel.org).
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SIPx (Open Source)
Asterisk is not really SIP or H.323
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one config file and mysql
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/
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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.
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©Stephen [email protected]
All the Vendors moving to SIP
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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 [email protected]
The impact of SIP providers of sip: accounts
Provide sip accounts like “hotmail” provides email accounts.
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Free World Dial (fwd) fwd.pulver.com
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www.atp.org (in Australia)
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And many many more, impossible to estimate the number
Providers of closed sip accounts (is this unproductive behaviour?):
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MSN
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Yahoo
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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 [email protected]
SIP based VoIP Carriers (too many to list)
ENGIN
AAPT
Internode/Agyle
ATP
ITouchTone
AOL
AT&T
BroadVoice
Broadvox Direct
Dialpad
Galaxy Voice
Global Village
G02Call
G02Call
iConnectHere
InPhonex
Lingo
Mutualphone
MyPhoneCompany
Net2Phone
Nikotel
NuFone
Packet8
QuantumVoice
SimpleTelecom
SIPphone
SIPphone
Skype (not SIP)
StanaPhone
SunRocket
TeleSIP
TeIIAX
TerraCall
USA Datanet
VoiceGlo
VoicePlus
VoiceWing (Verizon)
VoipJet
Vonage
VoxFlow
WebPhone
Yahoo
ZipGlobal
IIC (old ozemail)
There are some key questions to ask.
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Source: [email protected], 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 [email protected]
SIP based VoIP Carriers
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ENGIN, buy a black box from Dick Smith (no QoS).
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Internode combined with the Internode ADSL (has QoS).
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AU6c plus AU1c per minute to 90% of Australians, 25c/min to Mobiles.
Only available to AARNet Member Organisations.
Standard telephone rates
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Call other SIP addresses.
Call other VoIP Networks using an access code.
AARNet (with QoS)
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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
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AU10c (untimed) to any Australian number, AU29c/min to mobiles, free to another engin
user, AU3.5c/min to key international destinations.
around AU25c per local call,
Around AU X per minute for Long Distance.
©Stephen [email protected]
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 [email protected]
SIP will impact desk top Collaboration
Two problems seam to “dog” video conferencing, getting through firewalls and routing
PABX
SIP & H 323
Voice
GATEWAY
University
Network
H.323 Room
based Video
SIP & H.323
MCU
Desktop Video
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©Stephen [email protected]
SIP b2bua, and a
H.323 GK in Proxy
mode
AARNet
Internet with
QoS bandwidth
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 NAT’ing 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 [email protected]
What the customer wants?
Could Universities start loosing
their customers
to 3rd party providers?
Has this already started?
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©Stephen [email protected]
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
PSTN
Carrier
Voice
GATEWAY
SIP Proxy
Server
INVITE
Extn 3
575
E
INVIT
ed
Delay
St
ep
he
n.K
ing INV
ha ITE
m@
aa
Voice
Mail
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©Stephen [email protected]
Voice
GATEWAY
INVITE 1
1747
614194
rn
et.
PABX
ed
u.a
u
SIP UA
PSTN
Carrier
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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