SIP Tutorial Presenters: Stephen Kingham Stephen.Kingham@aarnet.edu.au And Prof Dr Quincy Wu (aka Aaron Solomon) solomon@ipv6.club.tw

Presenters: Stephen Kingham Stephen.Kingham@aarnet.edu.au VoIP Basics Presenters: Stephen Kingham Stephen.Kingham@aarnet.edu.au

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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 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 ©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 ©Stephen Kingham@aarnet.edu.au

VoIP Standards In 1995 we got the standard H.323. This is a Video Standard from the Carrier world and is based on ISDN. 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”). And we have some proprietary protocols/technology (read “painful”). ©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. ©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. ©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 A common thread for all successful VoIP and IP Telephony is the Voice expertise. The same can be said for the Video. ©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 ©Stephen Kingham@aarnet.edu.au

VoIP ©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 A common thread for all successful VoIP and IP Telephony is the Voice expertise. The same can be said for the Video. ©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 ©Stephen Kingham@aarnet.edu.au

IP Phone ©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. Important question to ask your IP Phone vendor in a multiple site environment: When I dial Emergency Services in one site which PSTN Gateway does the call go out. Now I dial the same Emergency Services number at another site – which Gateway? How hard is it to make the Emergency Services call go out the local PSTN Gateway 1st. New generation PBX programming is done using exception routing, some look more like programmes in perl or C. So a routing entry might look like this IF call from phone with IP Address 192.168.1.0/255.255.255.0 AND calling Emergency Services THEN send call to Gateway A first, Gateway B second IF call from phone with IP Address 10.10.1.0/255.255.255.0 AND calling Emergency Services THEN send call to Gateway B first then Gateway A IF call to Emergency Services (The default) ©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. A common thread for all successful VoIP and IP Telephony is the Voice expertise. The same can be said for the Video. ©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… A common thread for all successful VoIP and IP Telephony is the Voice expertise. The same can be said for the Video. ©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 Wu’s talk, www.apan.net Cairns 2004

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 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. ©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; } my $return = $stream -> find_vm_target; if ($return !~ /^200/) { print $return; $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); Slipper is an example of a modern commercial PABX Call Server up to even for a small Carrier ©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. ©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. ©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! ©Stephen Kingham@aarnet.edu.au

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 Skype (not SIP) StanaPhone SunRocket TeleSIP TeIIAX TerraCall USA Datanet VoiceGlo VoicePlus VoiceWing (Verizon) VoipJet Vonage VoxFlow WebPhone Yahoo ZipGlobal IIC (old ozemail) ENGIN, buy a black box from Dick Smith (no QoS). 10c (untimed) to any Australian number, 29c/min to mobiles, free to another engin user, 3.5c/min to key international destinations. Internode combined with the Internode ADSL (has QoS). 18c (untimed) to any Australian number, 30c/min to mobiles, free to another internode number, 15c/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 6c plus 1c per minute to 90% of Australians, 25c/min to Mobiles. Only available to AARNet Member Organisations. There are some key questions to ask. 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. ©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. ©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. ©Stephen Kingham@aarnet.edu.au

SIP will impact desk top Collaboration Two problems seam to “dog” video conferencing, getting through firewalls and routing ©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 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 Kingham@aarnet.edu.au

What the customer wants? Could Universities start loosing their customers to 3rd party providers? Has this already started? ©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. ©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 SIP enabled IP Telephony and VoIP and VIDEOoIP with Presence and Instant Messaging. Control handed back to the user, no more forwarding calls from one device to another. Mums and Dads and researchers are using the services from 3rd parties. Lots of hype and confusion in the market place. – watch out for destructive events (skype, and SPIT). All aspects of Telecommunications will be effected. SIP will bring mobility, freedom, cheap, and flexible. A new paradim in personal communication! Will have a larger Social and Economic impact than http? This community needs to work on SIP enablement of their infrastructure. Otherwise might University IT departments loose their users to a 3rd party? How then will you proceed?