Sidevrgud IRT 0020 loeng 12 15 nov 2005

Sidevõrgud IRT 0020 loeng 12 15. nov. 2005 Avo Ots telekommunikatsiooni õppetool, TTÜ raadio- ja sidetehnika inst. avo. ots@ttu. ee 148

SIP and Presence Converged Presence. Based Client Presence Services SIP Mobile Phone IP Phone Digital Phone Soft-phone Collaborative Applications IM Client PC w/ USB Camera 149

The Communications Detective • How Do I Contact Someone? – How can I contact them? • IM, voice, cell, e-mail, video? • Can my system talk to their system? – Are they available? • Are they on the phone or in a meeting? – Are they the right person for the inquiry? – How do I arrange a conference call? • What if they don’t dial into the teleconference? – How do I share information? 150

The Big Picture • Goal: a single communications infrastructure unifying collaboration, real-time communications, and presence – Convergence not of voice & data, but rather convergence all forms of real-time communications into a unified application • • Combined with collaboration tools Integrated into business applications & processes Extensible beyond organizational boundaries Delivering measurable business benefit by streamlining and improving the ability for individuals and groups to communicate • Converge non-real time communications (e-mail) – Voice but just one component of a converged communications & collaboration infrastructure 151

What is SIP? • Session Initiation Protocol protocol that establishes, manages (multimedia) sessions – also used for IM, presence & event notification – uses SDP to describe multimedia sessions • Standardized by IETF, 3 GPP (for 3 G wireless), Packet. Cable 152

Building Applications On Sip • Applications for example instant messaging can be built on SIP. Such applications are quick to write with any SIP toolkit and are easily extensible. invite OK User agent ACK OK OK Proxy server ACK LDAP server User agent lmn@tech. net--> 138. 47. 19. 103 153

Basic SIP message flow 154

SIP trapezoid outbound proxy SIP trapezoid a@foo. com: 128. 59. 16. 1 registrar 155

SIP event notification • Named events • Typically, used for events within conferences (“Alice joined”) and call legs (e. g. , call transfer) • Supports arbitrary notification bodies, typically XML SUBSCRIBE sip: alice@vmail. example. com SIP/2. 0 To: <sip: alice@example. com> From: <sip: alice@example. com>; tag=78923 Call-Id: 1349882@alice-phone. example. com Contact: <sip: alice@alice-phone. example. com> NOTIFY sip: alice@alice-phone. example. com SIP/2. 0 … Event: message-summary Subscription-State: active Messages-Waiting: yes Message-Account: sip: alice@vmail. example. com Voice-Message: 2/8 (0/2) 156

SIP event architecture • Does not try to route notifications (“application layer multicast”) as in SIENA – Filtering at PA under discussion (for low-bandwidth devices) • rate • content • But most ubicomp notification groups are probably small – and message volume not likely to provide much bandwidth saving via network-based filtering • Greatly simplifies trust model: no intermediaries that need to inspect content – can encrypt via S/MIME • However, can build redistribution “exploders” and list subscriptions (“subscribe to engineering@hp. com”) 157

SIP presence architecture REGISTER a@foo. com: 128. 59. 16. 1 SUBSCRIBE watcher PA NOTIFY Alice PUAs PUBLISH Bob <? xml version="1. 0" encoding="UTF-8"? > <p: presence xmlns: p="urn: …" entity="pres: alice@example. com"> <p: tuple id="sg 89 ae"> <p: status> <p: basic>open</p: basic> </p: status> <p: contact>tel: 09012345678</p: contact> </p: tuple> </p: presence> 158

Session mobility • Walk into office, switch from cell phone to desk phone – call transfer problem SIP REFER • related problem: split session across end devices – e. g. , wall display + desk phone + PC for collaborative application – assume devices (or standins) are SIP-enabled – third-party call control 159

Why Phone NUMBERS? • Users should not be burdened with device addresses, when it’s people they really care about • Addresses should be mnemonic and empower enterprises to manage the identities of their users sip: dbaron@mit. edu • It’s time to put E. 164 phone numbers behind us! • A. G. Bell did not say: “+1 -617 -252 -1232, come here. I need you!” 160

SIP. edu Architecture (Phase 1) SIP User Agent DNS SRV query sip. udp. bigu. edu DNS INVITE (sip: bob@bigu. edu) SIP Proxy INVITE (sip: 12345@gw. bigu. edu) SIP-PBX Gateway bigu. edu PRI / CAS PBX telephone. Number where mail=”bob” Campus Directory Bob's Phone 161

SIP. edu Architecture (Phase 2) SIP User Agent DNS SRV query sip. udp. bigu. edu INVITE (sip: bob@bigu. edu) If Bob has registered, ring his SIP phone; Else, call his extension through the PBX. bigu. edu DNS location DB SIP Proxy INVITE (sip: bob@207. 75. 164. 131) SIP Registrar REGISTER (Contact: 207. 75. 164. 131) Bob's SIP Phone 162

SIP. edu Call Flow Example • SIP DNS lookup for MIT. EDU points to SIP proxy – Sends INVITE to bob@mit. edu to proxy • SIP proxy checks MIT directory – Maps call to PBX extension – eg. 37669@mit. edu • SIP proxy checks dial plan – Routes call to PBX gateway • PBX rings phone 163

SIP. edu Configuration Internet SIP Server Campus Network SIP/PRI Gateway DNS Server LDAP Server PBX PSTN 164

DNS SRV Lookup DNS SRV Internet SIP Server Campus Network SIP/PRI Gateway DNS Server LDAP Server PBX PSTN 165

SIP INVITE SIP Internet SIP Server Campus Network SIP/PRI Gateway DNS Server LDAP Server PBX PSTN 166

LDAP Lookup Internet SIP Server LDAP Campus Network SIP/PRI Gateway DNS Server LDAP Server PBX PSTN 167

Call Sent to PBX Gateway SIP Internet SIP Server Campus Network SIP/PRI Gateway DNS Server LDAP Server PBX PSTN 168

Media Stream via Gateway to PBX RTP Internet SIP Server Campus Network SIP/PRI Gateway DNS Server LDAP Server PBX PSTN 169

SIP. edu Configuration Internet SIP Server Campus Network SIP/PRI Gateway DNS Server LDAP Server PBX PSTN 170

SIP to PBX and PSTN Calling Internet SIP Server Campus Network SIP/PRI Gateway DNS Server LDAP Server PBX PSTN 171

PBX and PSTN to SIP Calling Internet SIP Server Campus Network SIP/PRI Gateway DNS Server LDAP Server PBX PSTN 172

SIP Trunking Example Provider Gateway LA Provider Network SIP PRI PSTN PRI SIP Enterprise Location Provider Gateway NY IP-PBX New York Customer Location PBX Los Angels 173