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    of 35

    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
    3
    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 Stephen.Kingham@aarnet.edu.au 2006
    4
    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

    5
    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
    7
    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

    8
    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).

    Stephen Kingham@aarnet.edu.au

    9
    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

    10
    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

    11
    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 Kingham@aarnet.edu.au

    12
    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

    13
    VoIP
    PABX
    SIP Server
    or H.323 Gatekeeper
    Translate telephone numbers to IP addresses
    Other
    advanced IP
    network
    SIP & H323
    Voice
    GATEWAY
    AARNet
    Internet with
    QoS bandwidth
    SIP & H323
    Voice
    GATEWAY
    PABX
    PSTN
    Carrier
    Voice
    GATEWAY
    Stephen Kingham@aarnet.edu.au

    14
    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 Kingham@aarnet.edu.au

    15
    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

    16
    IP Phone
    SIP UA
    (IP Telephone)
    PABX
    SIP Server
    or H.323 Gatekeeper
    Translate telephone numbers to IP addresses
    SIP & H323
    Voice
    GATEWAY
    AARNet
    Internet with
    QoS bandwidth
    Stephen Kingham@aarnet.edu.au

    17
    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.
    Call Manager
    Server
    Site A
    PSTN
    Carrier
    Voice
    GATEWAY
    PSTN
    Carrier
    Voice
    GATEWAY
    Site B
    N
    O
    Calls to 000
    Stephen Kingham@aarnet.edu.au

    18
    Telephones in 2005+ The impact of SIP and
    3
    rd
    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 Kingham@aarnet.edu.au

    19
    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 Kingham@aarnet.edu.au

    20
    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
    21
    Also SIP Clients
    PDAs with SIP software
    MAC, Unix, and MSoft.

    Combination of Stephen Kingham and Quincy Wus talk, www.apan.net Cairns 2004
    Stephen Kingham@aarnet.edu.au

    22
    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

    23
    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;
    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;
    Andrew.Rutherford@iagu.net

    Slipper is an example of a modern
    commercial PABX Call Server up to even
    for a small Carrier

    24
    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

    25
    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

    26
    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

    27
    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
    There are some key questions to ask.
    Source: Wil.Daniels@acu.edu.au, Steering Committee Member for the AARNet IPTEL Working Group
    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)
    28
    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

    29
    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

    30
    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

    31
    SIP will impact desk top Collaboration
    Two problems seam to dog video conferencing, getting through firewalls and routing
    SIP & H.323
    MCU
    Desktop Video
    PABX
    SIP & H 323
    Voice
    GATEWAY
    University
    Network
    AARNet
    Internet with
    QoS bandwidth
    H.323 Room
    based Video
    SIP b2bua, and a
    H.323 GK in Proxy
    mode
    Stephen Kingham@aarnet.edu.au

    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 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.



    Stephen Kingham@aarnet.edu.au

    33
    What the customer wants?
    Could Universities start loosing
    their customers
    to 3
    rd
    party providers?

    Has this already started?
    Stephen Kingham@aarnet.edu.au

    34
    SIP UA
    SIP Proxy
    Server
    I
    N
    V
    I
    T
    E
    S
    t
    e
    p
    h
    e
    n
    .
    K
    i
    n
    g
    h
    a
    m
    @
    a
    a
    r
    n
    e
    t
    .
    e
    d
    u
    .
    a
    u
    PABX
    IN
    V
    IT
    E
    E
    x
    tn
    3
    5
    7
    5
    PSTN
    Carrier
    Voice
    GATEWAY
    IN
    V
    IT
    E
    6
    1
    4
    1
    9
    4
    1
    7
    4
    7
    1
    Voice
    Mail

    I
    N
    V
    I
    T
    E
    D
    e
    l
    a
    y
    e
    d
    PSTN
    Carrier
    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 Kingham@aarnet.edu.au

    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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