of 35 /35
SIP Tutorial Presenters: Stephen Kingham [email protected]. au And Prof Dr Quincy Wu (aka Aaron Solomon) [email protected]

voip-iptel-introduction-SRK1.ppt voip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-iptel-introduction-SRK1.pptvoip-ip

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SIP Tutorial Presenters: Stephen Kingham [email protected] Dr Quincy Wu (aka Aaron Solomon)[email protected]

mailto:[email protected]
mailto:[email protected]
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VoIP BasicsPresenters: Stephen Kingham [email protected]

mailto:[email protected]
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This work is the intellectual property of the author. Permission is granted for this material to be shared for non-commercial, educational purposes, provided that this copyright statement appears on the reproduced materials and notice is given that the copying is by permission of the author. To disseminate otherwise or to republish requires written permission from the author.

Copyright [email protected] 2006

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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

©Stephen [email protected]

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Other relevant talks at APAN Tokyo 2006• Monday 23 Jan

– SIP User Agents Configuration and Fault FindingSpeaker: Quincy Wu

– SER Configuration and SIP Peering including ENUMSpeaker: Stephen Kingham

– From Taiwan SIP Mobility in IPV4/IPV6 NetworkSpeaker:

– 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

©Stephen [email protected]

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What is IP Telephony, VoIP and VIDEO?

Presenter: Stephen [email protected]

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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

©Stephen [email protected]

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VoIP Standards1. 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”).

©Stephen [email protected]

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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.

©Stephen [email protected]

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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.

©Stephen [email protected]

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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

©Stephen [email protected]

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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

©Stephen [email protected]

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VoIP

PABX

SIP Serveror H.323 Gatekeeper

Translate telephone numbers to IP addressesOther

advanced IP network

SIP & H323 Voice

GATEWAYAARNet

Internet withQoS bandwidth

SIP & H323 Voice

GATEWAYPABXPSTN

CarrierVoice

GATEWAY

©Stephen [email protected]

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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

©Stephen [email protected]

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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

©Stephen [email protected]

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IP Phone

SIP UA (IP Telephone)

PABX

SIP Serveror H.323 Gatekeeper

Translate telephone numbers to IP addresses

SIP & H323 Voice

GATEWAYAARNet

Internet withQoS bandwidth

©Stephen [email protected]

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PABX IP Telephones : Emergency ServicesMake 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.

Call Manager Server

Site A

PSTNCarrier

Voice GATEWAY

PSTNCarrier

Voice GATEWAY

Site BCalls to 000

©Stephen [email protected]

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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.

©Stephen [email protected]

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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…

©Stephen [email protected]

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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 Wu’s 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 Wu’s talk, www.apan.net Cairns 2004

©Stephen [email protected]

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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.

©Stephen [email protected]

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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 -wuse 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

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The impact of SIP : SIP based PBXsSome 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.©Stephen [email protected]

http://www.openser.org/
http://yate.null.ro/pmwiki/
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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.

©Stephen [email protected]

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The impact of SIP providers of sip: accountsProvide 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!

©Stephen [email protected]

http://www.fwd.pulver.com/
http://www.atp.org/
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SIP based VoIP Carriers (too many to list)ENGINAAPTInternode/AgyleATP

ITouchToneAOLAT&TBroadVoiceBroadvox DirectDialpadGalaxy VoiceGlobal VillageG02Call

There are some key questions to ask.Source: [email protected], Steering Committee Member for the AARNet IPTEL Working Group

G02CalliConnectHereInPhonexLingoMutualphoneMyPhoneCompanyNet2PhoneNikotelNuFonePacket8QuantumVoiceSimpleTelecomSIPphoneSIPphone Skype (not SIP)

StanaPhoneSunRocketTeleSIPTeIIAXTerraCallUSA DatanetVoiceGloVoicePlusVoiceWing (Verizon)VoipJetVonage VoxFlowWebPhoneYahooZipGlobalIIC (old ozemail)

mailto:[email protected]
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VoIP Carriers that provide SIP: accountsProvide 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.

©Stephen [email protected]

http://www.fwd.pulver.com/
http://www.atp.org/
http://www.iptel.org/
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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.

©Stephen [email protected]

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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.

©Stephen [email protected]

http://www.voipsa.org/VOIPSEC
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SIP will impact desk top CollaborationTwo problems seam to “dog” video conferencing, getting through firewalls and routing

SIP & H.323MCU

Desktop Video

PABXSIP & H 323

Voice GATEWAY

UniversityNetwork

AARNetInternet with

QoS bandwidth

H.323 Room based Video

SIP b2bua, and aH.323 GK in Proxy

mode

©Stephen [email protected]

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32

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.

©Stephen [email protected]

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What the customer wants?Could Universities start loosing

their customers to 3rd party providers?

Has this already started?

©Stephen [email protected]

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34 SIP UA

SIP Proxy Server

PABX

PSTNCarrier

Voice GATEWAY

Voice Mail

PSTNCarrier

Voice GATEWAY

Someone calls 02 6222 3575

SIP FORKING (native to SIP)Never need to forward phones to other phones again!!!!This is a big mindset change for the user.

©Stephen [email protected]

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35

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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