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:presentity@pres.example.com SIP/2.0

Via: SIP/2.0/UDP watcherhost.example.com:5060

From: User

To: Resource

Call-ID: 3248543@watcherhost.example.com

CSeq : 1 SUBSCRIBE

Expires: 600

Accept: application/xpidf+xml, text/lpidf

Contact: sip:user@watcherhost.example.com

200 OK presentity->watcher 

SIP/2.0 202 Accepted

Via: SIP/2.0/UDP watcherhost.example.com:5060

From: User

To: Resource

Call-ID: 3248543@watcherhost.example.com

Cseq: 1 SUBSCRIBE

Expires: 600

Content-Type: application/xpidf+xml

Content-Length: 351

NOTIFY Presentity->watcher 

NOTIFY sip:user@watcherhost.example.com SIP/2.0

Via: SIP/2.0/UDP pres.example.com:5060

From: Resource    

To: User

Call-ID: 3248543@watcherhost.example.com

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:user1@user1pc.domain.com SIP/2.0 Via: SIP/2.0/UDP user2pc.domain.com To: sip:user1@domain.com From:sip:user2@domain.com;tag=ab8asdasd9 Contact: sip:user2@user2pc.domain.com Call-ID: asd88asd77a@1.2.3.4 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:user1@domain.com From: sip:user2@domain.com;tag=ab8asdasd9 Call-ID: asd88asd77a@1.2.3.4 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