Enterprise SIP Solutions
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Transcript of Enterprise SIP Solutions
Enterprise SIP SolutionsEnterprise SIP SolutionsVenkat R. Uppuluri — Director of Product Marketing
Enterprise SIP SolutionsEnterprise SIP Solutions
• SIP enables the convergence revolution– truly open standards based
• Presence & Instant Messaging and 3rd party call control– create a wealth of new services for enterprise end users – telephony-enabled address and buddy lists
• Advanced service creation using SIP– ad hoc video conferencing– user-customizable “find me” and “follow me”– user profiling– “here I am”
SIP-Enabled SolutionsSIP-Enabled Solutions
Voice MailServer SIP enabled
Network
PSTN
Internet
Router
Media Gateway
Call ManagementServer
SIPServer
ACDServer
SIP Enables InteroperabilitySIP Enables Interoperability
• Choice of vendors• Choice of products
– phones – call management servers– gateways– voice mail systems
• Choice of services– ACD Servers– contact center solutions
Analyst QuotationAnalyst Quotation
“Increasingly, the enterprise market seeks communications servers that support voice and data convergence, and that are highly scaleable. In addition, an ‘all-in-one’ hardware solution with a single software interface that is capable of managing voice and data communications leads to greater ease of implementation, management and potentially lower operational costs. Any solution that combines these features merits attention.”
— Jay Lassman, Research Director at Gartner
Standard Interfaces
Integrated Functions
InternetGatewayPBX
Converged SIP SolutionConverged SIP Solution
The SIP Convergence RevolutionThe SIP Convergence Revolution
• Simple provisioning with seamless mobility• IP Telephony and Video• Presence• Instant Messaging• 3rd Party Call Control
Simple ProvisioningSimple Provisioning
Simple Provisioning (continued)Simple Provisioning (continued)
Seamless MobilitySeamless Mobility
• Move your SIP Phone anywhere in network– no additional administrative work
• Register from anywhere– SIP Server becomes “virtual” PBX for
• for both local and remote users
• Address belongs to the user, not to devices– using one address, users..
• register multiple contacts • reached at preferred devices
IP Telephony features supported by SIPIP Telephony features supported by SIP
• Call Hold• Unattended Transfer• Unconditional Call
Forwarding• No Answer Call
Forwarding• Single-Line Extension• Incoming Call Screening• Call Pickup• Outgoing Call Screening• Call Return (*69)
• Music on Hold• Consultation Hold• Attended Transfer• Busy Call Forwarding• N-way Call• Find-Me• Call Park• Automatic Redial• Do Not Disturb• Message Waiting
Indicator
Revolution at the DesktopRevolution at the Desktop
• SIP applications– multimedia communications
• SIP is media agnostic– video is no different from voice
• SIP device is agnostic– phones– softphones– PDAs– tablet PCs– mobile phones
Presence and Instant MessagingPresence and Instant Messaging
• SIP for Instant Messaging and Presence Leveraging Extensions (SIMPLE)– several vendors who intend to implement SIMPLE– provides for presence and buddy lists– Instant Messaging in the enterprise– telephony enabled user lists
Presence Model Instant Message Model
PRESENCE UA
WATCHER UA
PRESENTITY WATCHER
PRESENCE SERVICE
SENDER UA INBOX UA
SENDER INBOX
INSTANT MESSAGE SERVICE
INSTANT MESSAGE
PROTOCOL
PRESENCE PROTOCOL
PRINCIPALS PRINCIPALS
Model for Presence and IMModel for Presence and IM
User AUser B
SUBSCRIBE
SUBSCRIBE SUBSCRIBE
NOTIFY
REGISTER
NOTIFY
SUBSCRIBE
NETWORK PROXY
SUBSCRIBE
SUBSCRIBE SUBSCRIBE
NOTIFY
REGISTER
NOTIFY
LOCAL PROXY
PROXY, REGISTRAR
PRESENCE AGENT
NETWORK PROXY PRESENCE
USER AGENTS
WATCHER
PRESENTITY
Presence and Instant MessagingPresence and Instant Messaging
PRESENTITY
WATCHER
Subscribe
202 Accepted
Notify
Presentity
200 OK
SUBSCRIBE watcher -presentity
SUBSCRIBE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP watcherhost.example.com:5060
From: User
To: Resource
Call-ID: [email protected]
CSeq : 1 SUBSCRIBE
Expires: 600
Accept: application/xpidf+xml, text/lpidf
Contact: sip:[email protected]
200 OK presentity->watcher
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP watcherhost.example.com:5060
From: User
To: Resource
Call-ID: [email protected]
Cseq: 1 SUBSCRIBE
Expires: 600
Content-Type: application/xpidf+xml
Content-Length: 351
NOTIFY Presentity->watcher
NOTIFY sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP pres.example.com:5060
From: Resource
To: User
Call-ID: [email protected]
CSeq: 1 NOTIFY
Content-Type: application/xpidf+xml
Content-Length: 352
SIP for Presence: Subscribe and NotifySIP for Presence: Subscribe and Notify
User 1
Message
200 OK
User 2
SIP for Instant Message: MessageSIP for Instant Message: Message
MESSAGE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP user2pc.domain.com To: sip:[email protected] From:sip:[email protected];tag=ab8asdasd9 Contact: sip:[email protected] Call-ID: [email protected] CSeq: 1 MESSAGE Content-Type: text/plain Content-Length: 29 My name is User1
SIP/2.0 200 OK Via: SIP/2.0/UDP user2pc.domain.com To: sip:[email protected] From: sip:[email protected];tag=ab8asdasd9 Call-ID: [email protected] CSeq: 1 MESSAGE Content-Length: 0
33rdrd Party Call Control: Basic Party Call Control: Basic
User A SIP Phone Controller Agent B PC
User PCUser B SIP Phone Agent A PC
HTTP GET
200 OK
INVITE SDP U1’
200 SDP A2
ACK SDP heldINVITE no SDP
200 SDP U1
ACK SDP A2’ACK
RTP
INVITE no SDP
200 SDP A1
Example of 3pcc: Click-to-DialExample of 3pcc: Click-to-Dial
Controller
Advanced Service CreationAdvanced Service Creation
• Ad hoc video conferencing• User customizable features
– find-me - follow-me– user profiling
• “Here I am”
Ad Hoc ConferencingAd Hoc Conferencing
• SIP enables ad-hoc conferencing of any media– audio– video – white board (T.120)– chat– media or applications yet to be defined (extensible)
• Advantage of SIP is that it’s standards based– conference using products from multiple vendors
• Invite people and add media at any time
Ad Hoc ConferencingAd Hoc Conferencing
Audio, Video,Instant Message
Audio, Instant Message
AudioSIP enabled
Enterprise Media Exchange
Audio, Video,Instant Message
Find Me – Follow MeFind Me – Follow Me
• Allows users to define – Who can reach them– Where they can be reached– When they can be reached – How calls are routed
• unconditionally or • based on a caller receiving
– no answer or– a busy signal
– What greetings are played for different callers
User ProfilingUser Profiling
• Users customize their own features• Set call handling rules
– ring specified device or all devices in unison or in sequence– forwarding rules
• when busy, no answer, redirection, unconditional to preferred destination or device
– rule activation• when on the phone
• timeout for incoming call
• identity of originating party
• the day of the week, a specific date or date range or holiday
• presence (specified in Client UI)
• time of day
• if internal or external call
• if voice or chat call
Here I AmHere I Am
• Find me & follow me– based on predefined rules
• Here I Am works on an ad-hoc basis – user logs in from any location
• indicates presence and manage calls from that location
– all communication now directed to that location• IM, chat, voice calls
SummarySummary
• SIP is open and extensible• Address belongs to the user, not to devices• Allows for better application integration and
enormous service creation opportunities • The Presence and Instant Messaging features in
SIP offer users great productivity gains• Enterprises are waiting for SIP based solutions
Quote from Quote from Burton GroupBurton Group“We've been telling our clients that SIP is the key to truly open IP telephony solutions but there have been very few solutions available for the enterprise. Zultys Technologies is among the first to offer a true SIP IP telephony solution with the features most enterprises require. Their product strategy is entirely based in SIP - starting with the MX1200 Enterprise Media Exchange - supporting voice, video, instant messaging, chat and presence capabilities that are interoperable with a range of other standards based SIP user agents and phones. The fact the MX1200 scales from 25 to 1200 users with fax, voice, video and data in one box, without additional hardware and only one software interface to manage it all, is also compelling for enterprise IT managers”
—Ray Kaneipp, VP of Networks & Telecom Strategy
Introducing the Introducing the MX1200MX1200• World’s first SIP-based Enterprise Media
Exchange– integrates voice, data, video, and fax
• MX1200 includes SIP– registrar and proxy– user agent client– user agent server– presence server– instant messaging
• Use standard SIP – desktop phone or – soft phone
• A truly open IP communications system!
www.zultys.com
ReferencesReferences
• RFCs and drafts– RFC 2778: A model for Presence and Instant Messaging– RFC 3261: Session Initiation Protocol– RFC 3248: SIP Extension for Instant Messaging– RFC 3361: Dynamic Host Configuration Protocol (DHCP-for-
IPv4) Option for SIP Servers– draft-ietf-simple-presence-09.txt– draft-ietf-sipping-3pcc-02.txt– draft-ietf-sipping-service-examples-03.txt
• Book– Understanding the Session Initiation Protocol, Alan B. Johnston