OCS-UC-SIP-OIP - Alphabet Soup Creates PBX-OCS "Co-existence Via Dual Forking"

Microsoft's OIP-Open Interoperability Program Pries Open the PBX

As the alphabet soup gets more letters, the complexity of the enterprise OCS-PBX user also warms up. The animated tutorial explains the details which can be found at:
ocs-co-existence.swf

In other to create not just better Unified Communications applications but unified switching now being called software powered voice, Microsoft created the OIP-Open Interoperability Program to:
- Develop Industry-Class Telephony Infrastructure that work seamlessly with OCS and Exchange UM-Unified Messaging
- Develop many solutions and new ideas
- Provide a forum for customers with setup, support, and use
- Test to enterprise-level standards for audio quality, reliability, and scalability
- Provide a means for scalable qualification of vendors
For more on OIP, go here: http://technet.microsoft.com/ucoip

Also explained in the animated tutorial is the OIP program designed to provide PBX implementation/integration in the following configurations:
1 - Standalone via gateway
2 - Standalone via direct SIP
3 - Co-existence via dual forking
- Direct SIP + PBX is qualified against Microsoft Dual-forking specification
Co-existence via dual forking - dual OCS phone and PBX phone
4 - Co-existence via dual forking with RCC-Remote Call Control
- PBX supports Dual forking plus RCC-Remote Call Control and
- CSTA-Computer Supported Telephony Application
Here's how it might work using RCC-Remote Call Control a user uses their OC-Office Communicator client for Presence/IM and uses the OC softphone to control their existing PBX phone. For example, a user checks the presence for someone via Office Communicator and then clicking on that user to call - but then having their PBX desk-phone call the number (and use that device). Remote Call Control can also be deployed in conjunction with Dual-Forking using the "dual" calling of simultaneous or sequential ringing feature of "forking" to call the phone(s).

First Party Call Control
Traditional telephony POTS, SIP and OCS are designed to provide for first party or first person call control. Examples of first party call control are:
- Make/receive or accept/answer incoming or outgoing calls
- Make another call to another user
- Conference both the callers

Third Party Call Control
Third party or third person call control is where another element, endpoint, server, telephone or device is involved in the call. Third party call control may mean that the endpoints share call control with another device such as a PBX, ACD-Automatic Call Distributor, CO-Central Office Switch or other device. The third party device such as a server may direct, redirect (fork) or disconnect the call.

SIP Forking Explained
See the animated tutorial for details
Parallel Forking is where the proxy forwards copies of the request to multiple destinations simultaneously. This is simultaneous ringing where the phone rings your work, home, cell and any other phone at the same time. Forking is the process of processing multiple requests such as desktop, softphone and cell phones or finding the first-available or other call center rules and the proxy is called a Forking Proxy. A stateless SIP proxy will act as a simple forwarding process - ex: Ethernet switch. A stateful SIP proxy will review, route, fork, and modify the SIP header thus staying involved with the SIP dialog until BYE - ex: a Firewall.

Sequential Forking is where the proxy forwards copies of the request to one target at a time and waits for a final response (or failure) before moving to the next address. This is sequential ringing where your work phone rings three times, then forks to your cell phone for three rings, then your home phone for three times and then forks to voice mail.

RCC-Remote Call Control also known as third-party call control is provided by CSTA-Computer Supported Telephony Applications. CSTA was developed by the European Computer Manufacturers Association (ECMA) and subsequently was formally standardized by the ITU-T, incorporating the Switch-to-Computer Applications Interface (SCAI). CSTA uses, among other technologies, SALT-Speech Application Language Tags specification and its SMEX-Simple Messaging Exchange element, telephony call control capabilities in MSS-Microsoft Speech Server to allow a developer to create sophisticated telephony-based speech applications that can exploit both basic call control services such as ANI-Automatic Number Identification (caller ID) and DNIS-Dialed Number Identification Service (800), using the included basic call controls, or extended call control services, to create custom call controls. CSTA uses ASN.1-Abstract Syntax Notation One - a notation system for describing data structures. ASN.1 while like programming language is in fact, not a programming language. ASN.1 is a flexible notation that allows programmers to define a variety of data types. ASN.1 is a set of encoding rules used to transform data into a standard format that can be decoded on any system that has a decoder based on the same set of rules.

SALT adds voice commands to web applications. SALT consists of a series of software applications meta-tags that are captured, interpreted and responds to audio-voice and DTMF-Dual Tone Multi-Frequency (touch tone) input. SALT is an extended set of markup (meta) tags based on XML-eXtensible Markup Language though compatible with HTML-Hyper-Text Markup Language and others. Many of the features of SALT include:

- Multi-mode browser-driven clients - PDA-Personal Digital Assistants, etc.
- Authoring Tools - IVR-Integrated Voice Response, grammar, dialogs, commands
- Speech-driven WEB pages with voice/data databases
- Interface to analog-digital telephones within SIP/VOIP servers and integrated voice recognition
- XML services with other business services such as SOAP to OLAP database connections
In the animated tutorial is an explanation of SALT in the OSI model.

In the animated tutorial are some of the basic SALT commands such as: - Prompt - Used to play audio prompts to direct a user to enter input - Grammar - Used to identify how audio sentences will be analyzed - Record - Used to trigger saving voice input

The concept of software powered voice is very compelling as you make voice just another XML service.

This presentation is included in online/onsite courses SIP Planning Guide and for OCS-101 Office Communications Server per person (volume and site license discounts available). For more information, go to:
http://www.techtionary.com/sip/planning-guide/

For customizing, special discounts, website animations, technical/sales training, technical writing and other services, go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com.

This tutorial is also included in TMC University special course on Microsoft OCS-Office Communications Server and Response Point at ITexpo.com. For more go here: http://www.tmcnet.com/voip/conference/west-08/tmc-university-microsoft-ocs.htm