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Get Smart in 2010 - Get this in your budget/plan.
OCS Forum Expo 2010 - Boulder, Colorado - June 15-16 at the St. Julien Hotel (www.stjulien.com) with keynotes from OCS MVPs, Media Gateway/SBC experts, customers, planners and others.
Attendees and exhibitors will receive thousands of dollars in valuable videos, training courses and online access to a "live" OCS system.
There will be sessions on OCS Planning, QoS, Security, Firewalls, Media Gateways, PBX Integration, Mobile OCS, Communicator, WAN-Bandwidth Planning, Customer Applications, Troubleshooting, Session Border Controllers and others.
Click here for prospectus, speaker/exhibitor/attendee information or contact Tom Cross cross@gocross.com or 303-594-1694.
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Here's some of the great OCS solutions presenting at OCS Forum Expo:
911 Enable Presents OCS Solutions at OCS Expo
911 Enable provides simple to deploy, easy to manage E911 solutions for IP telephony, including a solution designed to meet the unique requirements of Microsoft OCS. Its solutions include a national E911 call routing service, automated phone tracking appliance, and security desk notification system, which help organizations reduce liability concerns and meet E911 regulations. Visit www.911enable.com for more information.
AudioCodes Presents OCS Solutions at OCS Expo
AudioCodes is a leading manufacturer of Media Gateways for Microsoft Unified Messaging & Unified Communications, including Microsoft Office Communications Server 2007 R2. Utilizing AudioCodes Media Gateways, businesses are able to interface Microsoft communications applications to a wide range of TDM PBXs, IP-PBX, SIP Trunking and legacy PSTN trunking facilities. With unsurpassed voice quality, reliability, flexibility and scalability, AudioCodes Media Gateways have earned many accolades and should be your first choice when deploying Microsoft Unified Communications. Visit www.audiocodes.com/microsoft for more information.
NOVUS presents OCS Solutions at OCS Expo
Since 1983, Novus, LLC has been supporting manufacturers' reseller channels worldwide, serving over 15 countries on 4-continents. Engaged at the inception of VoIP, our thorough understanding of VAR's needs and end-user expectations has positioned Novus as the leading distributor of snom IP telephones deployed on Microsofts' OCS platform at www.novusllc.com
SNOM presents OCS Solutions at OCS Expo
snom technology is a global manufacturer of SIP-based IP telephones for business, carrier and high-end consumer markets. Based on open standards, feature-rich and affordable, snom's products are engineered to fulfil their vision of ubiquitous, standards-based VoIP/SIP/OCS. Based in Berlin, snom's sales/distribution network extends to over 40 countries at www.snom.com
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Get Smart in 2010 - Get this in your budget/plan.
OCS Forum Presents - OCS R2-2010-Ultimate Course
5-Day Training - "Hands-on" Labs with 700+ page manual
Critical Course for Planning OCS Design & Certification Study for Exam 70-638, Exam 70-262 and other tests
OCS Forum (http://www.ocsforum.com) announced its new OCSR2-2010 Ultimate course. "R2 and Wave 14 coming in 2010 confirms Microsoft place in the new telecommunications networking business," noted Tom Cross OCS Forum CEO. "R2 is having a significant impact on corporate voice telecommunications strategies indicating the end of the TDM-time division multiplexed PBX-Private Branch eXchange systems is now insight. Microsoft is also driving companies like Cisco, Nortel, Avaya, NEC, Mitel, ShoreTel to rethink their featuresets and capabilities because while R2 is a new game when or Wave 14 is released in 2010, their days are numbered," Cross commented.
This five -day (5-day) "hands-on" lab course with 700+ page manual focuses on the core components of OCS 2007 R2, including:
- Instant Messaging (IM) between everyone in the organization in the office or remote
- Application and Desktop sharing for true collaboration
- Audio/Video Conferencing including internally hosted audio conference calls and Live Meetings
- Integration with Exchange Server 2007 Unified Messaging
- Securing the environment to protect communications
The course provides many hands-on labs to practice and reinforce learning of many new concepts. After completing this course, students should be able to design, install, configure, maintain, monitor, and troubleshoot the core components of OCS 2007 R2.
Up to $6,000 in discounts are available for 2010 classes if scheduled before December 1. Call Tom Cross at 303-594-1694 or cross@gocross.com for details. |
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Part 2 of 2 - Parallel & Sequential Forking with 1st & 3rd-Party Call Control with RCC
1st Party Call Control - traditional telephony POTS, SIP and OCS are designed to provide for first party or first person call control.
3rd Party Call Control - 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.
Key Point - Forking is critical to advanced SIP features such as "find-me follow-me."
- Parallel Forking - the proxy forwards copies of the request to multiple destinations simultaneously.
- Sequential Forking - 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.
Critical to this process is ringing to let the caller alert the callee or (called party) of the call. In traditional PSTN-Public Switched Telephone Network communications "early media" refers to ringing and announcements to indicate the status of the call - ringing, busy, fast busy, call redirection, status "you have 12 callers ahead of you." In SIP, the forking process provides "early media" sending specialized ring-tones, audio announcements (e.g. call center status announcements), images or video before SIP session is accepted. However, there is NO common means of providing signaling to the receiver because of different types of hardware, softphones, UI-user interfaces, ringing devices and many other factors. In addition, early media may be onmidirectional or sequential forking, bidirectional or parallel-dual forking.
Early media failures can occur when the callee picks up and the UAS sends a 200 (OK) response with an answer, in parallel with the first media packets. If the first "early media" packets are received by the caller - UAC-User Agent Client, "media clipping (at the beginning of the media sequence)" or "media dipping" (at the end of the media sequence) can occur. This can occur is that the UAC cannot send media until the 200 (OK) response from the UAS arrives. Causes for clipping can be manyfold, however, UAC signaling, packet arrival delays), bandwidth, different SIP "methods" and commands between SIP systems and other factors. In addition, SIP signaling can typically take a different routing path than the media (communications) transmission which can be one of the factors causing media clipping. "Late media" announcements (such as "will you take a survey" or (click for special offer") occur after the BYE and may have the same problems. Details can be found by reading RFC 3260. In summary, incompatibility between SIP systems can result in communications chaos.
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 is an OSI protocol stack that provides an open system interface to a PBX-Private Branch eXchange, ACD-Automatic Call Distributor or CO-Centrex central office switching. 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.
Next week's OCS tutorial - "XMPP-Super-sized Presence."
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This is the text from Part 1 of 2 - OCS-UC-SIP-OIP - Alphabet Soup Creates PBX-OCS - "Co-existence Via Dual Forking" - Microsoft's OIP-Open Interoperability Program Pries Open the PBX
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, click here.
Also explained in the animated tutorial, 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). |
| Designing and Planning Scalable SIP Networks
- New Enhancements to SIP2010 Planning Guide Course
The complexity of SIP-Session Initiation Protocol networks is increasing geometrically. As with any innovation new developments are required for their widespread diffusion. As most of you know, the incredible simplicity of SIP is device independence with direct communications between endpoints. The proliferation of SIP will grow even more from machine-to-machine communications with remote endpoints solving and fixing problems without human intervention. However, one of the challenges in designing a SIP network is the use of various proxy servers to facilitate security, trunking, routing, applications and other functions. Tekelec issued a very interesting white paper on SIP Signaling Router Application Handbook. While the white paper did not address SIP proxy security (I did write about it in an earlier blog item called, The Many Flavors of SIP Trunking Solutions for SIP Systems & OCS-Office Communications Server.) Needless to say, the paper provides an excellent foundation for designing complex, scalable and future-ready networks. The five areas of discussion presented (with modifications I added) are:
1 - SIP Signaling-Server Virtualization
2 - SIP Trunking
3 - SIP Number Transparency
4 - SIP Routing
5 - SIP Proxy Peering Networks
Here are the highlights in text format:
1 - Benefits of Server Virtualization aka Abstraction
- Distributed user endpoints and application servers
- Scalable growth with QoS control abstraction
- MACS* via signaling "abstraction"
*Moves, Adds, Changes & SAS-Stand Alone Survivability - SAS enables backup for SIP devices by the multiple "abstract" local or cloud Media Gateway(s)
2 - Benefits of SIP Trunking
- Session and Signaling control layers for:
- On-net (IP PBX to IP PBX)
- Off-net (IP PBX to local PSTN)
- Off-net (IP PBX to LD/IDDD PSTN)
- SPOC/SMOC-Single/Multiple Point of Connection
- National control points for access (vendors) - Migration path to ALL-SIP & IMS-3GPP
3 - Benefits of Number Transparency
- End-to-end media control (transparency) for voice or other media types
- Access to SS7 applications for text messaging, mobility or call center routing
- Enterprise control over call completion whether PSTN or SIP rather than provider
- Migration path to E.164 & IMS-3GPP
4 - Benefits of SIP Routing
- Eliminates "mesh" network mess
- Add intelligent Session Layer 5 routing - Maintains end-to-end media transparency
- Add media CoS-Classes of Service - Adds SPOC NOC-Network Operations Center
- Migration path to ALL-SIP & IMS-3GPP
5 - Benefits of SIP Proxy Peering Networks
- Platform for multi-vendor services
- Ensures multiple CoS & QoS options
- Migration path to ALL-SIP & IMS-3GPP
- Foundation for any future network needs
click here to view tutorial
This new section is now part of the enhancements to SIP2010
2-5 day comprehensive SIP training. For a complete outline and details, click here.
Get Smart in 2010 - Get this in your budget/plan.
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Here's the Course Outline and TMCU OCS Certification Testing:
1 - Intro, Overview and Future Outlook of OCS - The Role of the PBX in an OCS World
2 - Demonstration of Key Features in OCSR2 and "Live" Demonstration
3 - New Features in R2 and Coming Soon in 2010
4 - OCS Planning Guide
5 - Edge Server Planning Guide
6 - TMU Certification Exam
ITEXPO is the event with an educational program that teaches resellers, enterprises, SMBs, and Government Agencies how to select IP-based voice, video, fax, and unified communications to purchase or resell. It's where service providers learn how to profitably roll out services their subscribers are clamoring for. ITEXPO is where buyers, sellers, resellers, and manufacturers meet to forge relationships and close deals.
Get serious or at least smart - read new book "Facebook Marketing for Dummies - click here
Cool Microsoft Blog on OCS for Educational Community - click here |
| New OCS Forum Job Board - click here
-- New Jobs from Gold Systems & Level3
Post your UC, UM, LCS, OCS, PBX, Integration and related jobs on OCS Forum Job Board at no charge.
Premium advertising available.
Send your positions to cross@gocross.com. |
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CrossTalk Named One of the Top-10 Telecommunications Blogs
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All videos are FREE to OCS Forum 2010 Expo attendees and exhibitors.
Previous Videos
- OCS Front End Server
- OCS Planning Tool
- Edge Planning Tool
- Getting To Know Microsoft Office Communicator
- OCS User Creation
- Mediation Server
- Edge Server |
Free Technical "Just Enough Just-in-Time" Knowledge from:
The World's First and Largest Animated Library on Technology with more than 3,000 animated tutorials.
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OCS Forum provides classroom and webseminar training as well as a non-production environment for those IT departments without additional equipment, budget or time. This allows planners and users to test ideas, dial in and dial back out, IM file transfers, remote desktop sharing, video conferencing, run scenarios, review logs, break linkages and learning about new telephony features and network access. OCS Forum is also designed for both the system integrator/consultant who wants to learn about OCSR2 without having to build their own system as well as the enterprise customer who doesn't have the time, resources or knowledge to develop one. OCS Forum Labs are designed to be "hands-on" or "over-the-shoulder" with experts available for Q&A and classes for feature-specific review.
About OCS Forum
OCS Forum is a vendor-independent laboratory environment designed for learning, technical guides, knowledge resources and online "live" services. OCS Forum provides planning, project management, consulting, training, case studies, white papers, speaking engagements, market/customer research, network planning and other services.
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