148 sidevõrgud irt 0020 loeng 1215. nov. 2005 avo ots telekommunikatsiooni õppetool, ttÜ raadio-...
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SidevõrgudIRT 0020
loeng 12 15. nov. 2005
Avo Otstelekommunikatsiooni õppetool,
TTÜ raadio- ja sidetehnika [email protected]
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Digital
PhoneIP Phone
SIP
Presence
Services
Soft-phone IM Client PC w/ USBCamera
Converged Presence-
Based Client
Mobile
Phone
Collaborative
Applications
SIP and Presence
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• 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?
The Communications Detective
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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
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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, 3GPP (for 3G wireless), PacketCable
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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.
Useragent
Proxyserver
Proxyserver
User agentACK
OK
invite invite
OK
ACK
invite
OK
ACK
LDAPserver
LDAP server
[email protected]> 138.47.19.103
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Basic SIP message flow
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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:[email protected] SIP/2.0To: <sip:[email protected]>From: <sip:[email protected]>;tag=78923Call-Id: [email protected]: <sip:[email protected]>
NOTIFY sip:[email protected] SIP/2.0…Event: message-summarySubscription-State: active
Messages-Waiting: yesMessage-Account: sip:[email protected]: 2/8 (0/2)
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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 [email protected]”)
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SIP presence architecture
PA
[email protected]: 128.59.16.1
watcher
PUAs
Alice Bob
PUBLISH
REGISTERSUBSCRIBE
NOTIFY
<?xml version="1.0" encoding="UTF-8"?><p:presence xmlns:p="urn:…" entity="pres:[email protected]"><p:tuple id="sg89ae"> <p:status> <p:basic>open</p:basic> </p:status> <p:contact>tel:09012345678</p:contact></p:tuple></p:presence>
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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 stand-ins) are SIP-enabled
– third-party call control
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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 userssip:[email protected]
• It’s time to put E.164 phonenumbers behind us!
• A.G. Bell did not say:
“+1-617-252-1232, come here. I need you!”
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SIPProxy
DNSSIP-PBXGateway
PBX
INVITE (sip:[email protected])
INVITE(sip:[email protected])
DNS SRV query sip.udp.bigu.edu
telephoneNumberwhere mail=”bob”
PRI / CASbigu.edu
CampusDirectory
SIP User Agent
Bob's Phone
SIP.edu Architecture (Phase 1)
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DNS
INVITE (sip:[email protected])DNS SRV query
sip.udp.bigu.edu
bigu.edu
SIP User Agent
SIP.edu Architecture (Phase 2)
locationDB
If Bob has registered, ring his SIP phone; Else, call his extension through the PBX.
REGISTER(Contact: 207.75.164.131)
INVITE (sip:[email protected])
SIPProxy
SIPRegistrar
Bob's SIP Phone
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SIP.edu Call Flow Example
• SIP DNS lookup for MIT.EDU points to SIP proxy– Sends INVITE to [email protected] to proxy
• SIP proxy checks MIT directory– Maps call to PBX extension – eg. [email protected]
• SIP proxy checks dial plan– Routes call to PBX gateway
• PBX rings phone
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PBX PSTN
SIP/PRIGateway
SIP.edu Configuration
SIP Server
Campus Network
Internet
DNS Server
LDAP Server
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PBX PSTN
SIP/PRIGateway
DNS SRV Lookup
SIP Server
Internet
DNS Server
DNS SRV
LDAP Server
Campus Network
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PBX PSTN
SIP/PRIGateway
SIP INVITE
SIP Server
Internet
DNS Server
SIP
LDAP Server
Campus Network
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PBX PSTN
SIP/PRIGateway
LDAP Lookup
SIP Server
Internet
DNS Server
LDAP
LDAP Server
Campus Network
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PBX PSTN
SIP/PRIGateway
Call Sent to PBX Gateway
SIP Server
Internet
DNS Server
SIP
LDAP Server
Campus Network
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PBX PSTN
SIP/PRIGateway
Media Stream via Gateway to PBX
SIP Server
Internet
DNS Server
RTP
LDAP Server
Campus Network
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PBX PSTN
SIP/PRIGateway
SIP.edu Configuration
SIP Server
Internet
DNS Server
LDAP Server
Campus Network
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PBX PSTN
SIP/PRIGateway
SIP to PBX and PSTN Calling
SIP Server
Internet
DNS Server
LDAP Server
Campus Network
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PBX PSTN
SIP/PRIGateway
PBX and PSTN to SIP Calling
SIP Server
Internet
DNS Server
LDAP Server
Campus Network
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SIP Trunking Example
IP-PBX
Enterprise Location
Provider Network
SIPProvider Gateway
NY
New YorkPBX
Customer Location
Los Angels
SIP
Provider Gateway
LA
PSTN
PRI
PRI