Free Training for Channel Partners
May 12, 2008
TECHtionary.com Announces Free Training for Channel Partners
on SIP, OCS-Office Communications Server, Digital Communications Courses
Courses Proven to Reduce Sales Costs & Buy Cycle
TECHtionary.com today announced new program to help channel partners (independent, agents, VARs, dealers, brokers, interconnect) grasp new technologies faster and increase revenues faster. These free courses “gives access to channel partners who are time-constrained, mobility-driven and performance-pressured in today’s ruthless business world. Channel professionals can also learn on the go and anytime they want to whether waiting at the airport or a customer’s office, driving down the street or riding on the bus/train/plane,” Cross added.
Courses are free for agents of TECHtionary-TBI-Telecom Brokerage Inc. where agents not only receive free training but commissions on network services delivered to customers. For more information on this program, click here: http://www.tbicom.com/tbiu/. There is no limit to the number of employees who can attend the classes though each employee must be registered with TBI. TBI-Telecom Brokerage Inc, is one of the largest master agency in the U.S. TBI represents all of the leading network services providers with a complete set of Local, LD, Internet, Data and advanced telephony SIP products. According to Geoff Shepstone, CEO of TBI, “Tom Cross is exceptionally technically astute - the most technically proficient individual I know of in the industry. Yet he has the rare ability to deliver the message in a way the laymen can understand. These courses can bring immediate returns to channel partners.”
Course titles include:
- CTM-Communications Technology Manager -
Introduction to the Fundamentals of Digital Communications Technology
- WTM-Wireless Technology Manager
Job Training and Implementation of WiFi, WiMax, Cellular and IMS
- VBE - VoIP Business Executive
Channel Provider & Partner Business Sales & Technical Strategies for VoIP-SIP
- STE - SIP Essentials - Technology Business Executive
SIP Essentials - Comprehensive guide SIP-Session Initiation Protocol
- NDP-Network Design Professional
Planning for optical, routing, routing protocols and preparation for industry-wide Cisco, Microsoft, Avaya, Comptia and other certifications
- OCS-Office Communications Server Complete
Indepth explanation of Microsoft OCS-Office Communications Server
- Media Library
Audio-video tutorials are available in number of media formats such as Adobe-Flash (.swf), .mp3 (audio only), iPod (.m4v) and Apple QuickTime (.mov).
For more information please call Tom Cross at 303-594-1694 or cross@gocross.com.
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Voice Peering Fabric – The Future of SIP Interconnections
May 6, 2008Imagine that your corporate ethernet LAN-Local Area Network and any other Fortune 500 company LAN can connect via a Voice Peering Fabric (www.thevpf.com), a service of Stealth Communications. Calls from any LAN business or otherwise can be routed through the VPF and then to your LAN and to your desk. Look Ma, no Internet routing hops would be involved. Any SIP provider or Enterprise can setup peering through this system via ethernet. The calls would cost nothing per minute, aside from the fixed monthly cost of the digital link. This concept just extends LAN-to-LAN functionality to almost anywhere. VPF is accessible at major carrier hotels in North America and London, UK. If an Enterprise or Service Provider is not located at a common location as the VPF, they can provision Ethernet connectivity from their location to the nearest VPF location.
According to Shrihari Pandit, CEO of the Stealth Communications, “we created a program called the VPF Carrier Alliance to assist organizations in obtaining competitive Ethernet connectivity into the VPF. The alliance is composed of regional, national and international carriers that specialize in providing Ethernet connectivity. Once an organization is connected into the VPF, they are able to establish SIP peering with other connected organizations.” Members of the VPF use the service to buy and sell voice origination and termination services, as well as ASP services while providing complete transparency. The service allows organizations to instantly identify all available networks and services at the most competitive rates. Members then get connected to these networks and services within minutes by using the "fabric" component of the VPF. In addition, organizations can access the VPF ENUM and SRV Registry, as a toll bypass system, to send and receive free telephone calls across the VPF with other users of the registries. Here are some of the services offered by members of the VPF:
- SIP/H.323 Origination and Termination Trunks
- Operator Service, Directory Assistance & E911 Routing and
- Telco Database Services: 8XX, CNAM, LIDB, LNP.
In the case a company wants to exchange voice traffic with another company that is not on the VPF, they can either peer with that company over another network (such as the public Internet) or route that call through the PSTN (Public Switched Telephone Network.) In the event the call needs to be routed to the PSTN, there are carriers on the VPF that can route such traffic to the PSTN on their behalf (bilateral relationship.) VPF supports bilateral and multilateral peering relationships. Bilateral peering is a term that describes when two parties decide to exchange traffic on a settlement basis. For example, there may be a cost per minute to send calls. Multilateral peering enables a community of companies to exchange traffic for free (without settlement) between members of the community. Multilateral peering typically uses ENUM (Electronic Number Mapping) that maps telephone numbers to Internet addresses and is based Internet DNS technology. For more go here: http://www.thevpf.com/
If you want some real “live” peering, plan to attend the Voice Peering Forum, a biannual conference, scheduled for June 23-24 2008 at Hotel Nikko in San Francisco which brings together over one hundred unique organizations from all segments of the information technology and telecommunications industry to network and discuss the latest in peering, routing and interconnection of networks and the applications they support. Go here to get more http://www.voicepeeringforum.com/
If you want to know more, this information is part of OCS-101 and SIP Essentials 2.0c courses available onsite and online. The online version is $299 for SIP 2.0c and $499 for OCS-101 Office Communications Server per person (volume and site license discounts available). For more information go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com.
Discounts are also available to members of the SIP Forum and MS Partners for $99 per student during May.
Related Tags: voice peering, ethernet connectivity, multilateral peering, peering, services, organizations
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OCS Exposed - SQL-Structured Query Language for Telephony
OCS Exposed - SQL-Structured Query Language for Telephony
An internal telephony user can connect to another internal telephony user by using the Office Communicator 2007 client. A user can initiate a call to another user by either selecting the user from a contacts lookup list or dialing that user’s contact number. When the user initiates a call using a SIP-Session Initiation Protocol client, the client sends an SIP "Invite" message along with the SIP URI-Uniform Resource Indicator of the call recipient to the Front End Server. After receiving the SIP Invite, the Front End Server queries the SQL Server 2005 database to check if the SIP URI for the call recipient is present in the database. If the SIP URI is available, the Front End Server applies client rules on the Invite, and then routes the Invite to all active SIP clients corresponding to the URI of the call recipient. If the Front End Server receives a busy or does not receive an acknowledgement from any one of the SIP clients of the call recipient, the Front End Server then routes the call to the Exchange Server 2007 voice mail service.
The voice mail service generates an e-mail to the call recipient along with the voice mail attachment as an audio file. However, if an acknowledgement is received from any one of the SIP clients of the call recipient, the Front End Server responds with a Ringing message to that client. The Front End Server also sends a Cancel Invite message to all the other registered clients of the call recipient. A call session is established when the call recipient answers the call by using the SIP client. The Front End Server then opens a media stream between the clients of both users. After the conversation, either of the clients can send a Bye message and the Front End Server terminates the call session.
The voice mail service generates an e-mail to the call recipient along with the voice mail attachment as an audio file. However, if an acknowledgement is received from any one of the SIP clients of the call recipient, the Front End Server responds with a Ringing message to that client. The Front End Server also sends a Cancel Invite message to all the other registered clients of the call recipient. A call session is established when the call recipient answers the call by using the SIP client. The Front End Server then opens a media stream between the clients of both users. After the conversation, either of the clients can send a Bye message and the Front End Server terminates the call session.
The Front End Server also indexes the SQL Server 2005 database to translate (map-resolve) the normalized (canonical) number to a user URI-Uniform Resource Indicator. Note: URI refers to the complete SIP telephone address not just the mail URL-Uniform Resource Locator. If the Front End Server does not receive an acknowledgement from any one of the SIP clients of the call recipient, the configured InfoAgent or Outbound Router logic running on the Front End Server detects that the call recipient is not answering the call.
For those unfamiliar with SQL, SQL-Structured Query Language which is an ANSI-American National Standards Institute standard computer language for accessing and manipulating database systems. SQL statements are used to retrieve and update data in a database. SQL Statements contain "same type" information (e.g. store address, SKU, shelf count) sometimes referred to as columns. The DML-Data Manipulation Language is used to retrieve, add/insert and change/modify database information.
For a detailed animated tutorial on SQL, database concepts and other topics, go to http://www.techtionary.com
If you want to know more, this information is part of OCS-101 and SIP Essentials 2.0c courses available onsite and online. The online version is $299 for SIP 2.0c and $499 for OCS-101 Office Communications Server per person (volume and site license discounts available). For more information go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com.
Discounts are also available to members of the SIP Forum and Microsoft-MS Partners.
Related Tags: query language, structured query, recipient front, uniform resource, Server, server
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MPLS-Multi-Protocol Label Switching Multimedia Presentation
May 2, 2008To begin with, IP-Internet Protocol packets may have a number of labels or "tags" attached to them. MPLS-Multi-Protocol Label Switching is just one type of label. In a Provider Provisioned Virtual Private Network known as PWE3 or PPVPN, there may be more than one label. Here are some terms associated with labeling:
- Push - add a label
- Swap - replace the label
- Pop - remove the label
Here are some other terms associated with labeling.
The outer label identifies the LSR-Label Switch Router.
The inner label identifies the destination VPN-Virtual Private Network.
Shown in the audio-visual tutorial is the IP-Internet Protocol packet before and with the MPLS “label” attached or “tagged” on as it was originally called. MPLS consists of four elements, label bits, experimental bits, a stack bit and TTL-Time-To-Live bits which indicate the number of Label Switch Routers passed. Shown here is the “multi-protocol” part of MPLS and how it works with the other major networking protocols such as ATM, Frame Relay, Ethernet and others.
As if one label was not enough, MPLS providers may add more labels. These labels may exist within the MPLS provider’s network but may be removed or "popped" as they leave the network to the customer premise or "edge" or LER-Label Edge Routers. A PPVPN control module adds "pushes" labels and determines routing via LSR-Label Switch Routers where labels may be "swapped" as they change or cross to other networks called AS-Autonomous Systems. The term LVC-Label Virtual Channel has been associated with this emerging concept.
As long as each MPLS provider or AS-Autonomous System communicates the value of QoS-Quality of Service for the MPLS label to other MPLS providers and routes it accordingly, each carrier can determine their own MPLS labeling system. That is, if each AS carrier routers video as video or email as email or other known rules, then the packets will be treated with the desired QoS. When leaving the MPLS Network or network "edge" the MPLS and other label(s)
are popped (off) and the IP packet returns its original size.
Related Tags: multi protocol, label switch, label, Label, network, protocol
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Plan to Attend - Introduction to OCS & Internet Telephony Expo - Tuesday - September 16 - Los Angeles Convention Center
April 24, 2008Mark your calendar and plan to TMC University’s one-day
Introduction to OCS – Tuesday – September 16 - Los Angeles Convention Center
for details go here:
I look forward to teaching the class and if you have an specific questions, please let me know.
If you want to get a "head start" on your learning, this information is part of OCS-101 and SIP Essentials 2.0c courses available onsite and online. Th online version is $299 for SIP 2.0c and $499 for OCS-101 Office Communications Server per person (volume and site license discounts available). For more information go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com.
Discounts are also available to members of the SIP Forum.
Related Tags: available
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Virtualization – “You Haven’t Lost Your Mind, Just the Data”
April 20, 2008SIP/OCS and related applications such as conferencing, IM, video and other CPU-intensive systems would be many of the key applications to “outsource” to virtualization. Virtualization has been and being expanded in corporate environments as complex applications and user needs increased additional server systems are required. CPU-Central Processor Virtualization or Virtualization allows one computer system to emulate as multiple “virtual” computer systems. A VMM-Virtual Machine Monitor also known as a hypervisor is a system of system designed to control execution processes of multiple guest operating systems on a single machine. The term host refers to the execution context of the VMM. World switch refers to switching or movement between host and guest. Guest control is the appearance by the user of complete control over all machine systems such as memory, CPU and other peripheral devices (disk, tape, communications, etc.).
In a distributed network architecture, multiple computer servers can be combined into one "virtualized" system operating (running) different applications, operating systems and network connections. Virtualization can delegate IT-Information Technology tasks such as server consolidation/decentralization, communications gateways, enhanced distributed processing such as graphics rendering, legacy migration and security. VMM-Virtual Machine Monitors is a sub-layer of system software which enables multiple VM-Virtual Machines to share the same hardware platform and allows applications to run without modification. Here are some of the key capabilities of VMM Hardware: 1) workload isolation, 2) workload consolidation, 3) workload embedding and 4) workload migration. Virtualization is not just a data processing issue. SAN-Storage Area Network architecture is moving from data storage to OBS-Object-Based Storage. In an OBS system data is stored as an object not as data blocks with the data (metadata) more in the form of an application that stores and retrieves data defining objects or content. Metadata (data about data) or the means of data organizing is one of the critical innovations required. Metadata is the algorithms (rules-of-thumb) or attributes the OBS uses to determine the object and its history. Metadata can be distributed to search engines, authentication/authorization/identification, regulatory compliance requirements and other applications. In some cases, entire libraries of data are organized into CAS-Content Addressable Storage to reduce over duplication and increase efficiency in the backup system. OCS or home users could create virtual “partitions” isolating multiple user environments such as adding (dedicating) additional resources to a PC game, SIP, IM, communications or entertainment systems personal video recorder-type environments. In addition, virtualization can isolate (partition) incoming threats such viruses or spyware.
If you want to know more, this information is part of OCS-101 and SIP Essentials 2.0c courses available onsite and online. Th online version is $299 for SIP 2.0c and $499 for OCS-101 Office Communications Server per person (volume and site license discounts available). For more information go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com.
Discounts are also available to members of the SIP Forum.
Related Tags: virtualization, system, systems, Virtualization, applications, virtual
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OCS-Office Communications Server Exposed – Microsoft Media Stack
Among the many ways OCS improves on voice quality is via:
1 - QoS-Quality Controller
OCS provides for "dynamic adaptation" to real-time network conditions with progressive bit rate adjustments and insures that no voice sessions are dropped during this process. However this does not prevent content loss or jitter. According to various sources, even a 1% loss can significantly degrade the user experience with G.711, which is the standard for toll quality for voice CODECs. That is, different CODECs (e.g. G.711/729) may be used at different locations within the same company and more likely between different companies. MOS-Mean Opinion Scores reduces rapidly with each time a voice conversation is processed by a CODEC. Whether you call it Transcoding or Tandem Encoding is a network element to be standardized when ever possible.
2 - Voice Activity Detection/Silence Suppression
Voice clipping, chopping or dipping occurs as the result of the VAD-Voice Activity Detector. VADs are used for silence suppression in packet voice systems, due to the need not to use bandwidth to send packets when there is no voice; think of this as not sending silence. If below the signal threshold, then the media is clip/cut as it is perceived as noise not voice. This is called "front end speech clipping." "Dipping" is clipping at the end of the voice segment. VAD also reduces or suppresses echo suppression in echo cancellers.
VADs are also used to reduce room or background noise in IP phones, speakerphones and microphones. VAD systems determine the difference between human voice, tones, unvoice, "white noise" (similar in concept to white light (composed of equal amounts of all visible light frequencies) - a sound composed of an equal mix of all audible frequencies) and comfort noise (noise generated to let the user know the call has not been disconnected) and other sources. There are three or more types of noise or non-noise detectors: known energy level, adaptive energy level and spectral energy based on compression.
3 - Audio Optimization
Advanced voice engineering is used to enhance noise suppression, AGC-Automatic gain control, automatic echo cancellation and other techniques. Echo is caused by three principle factors: talker echo, listener echo and loss of interaction (human and cultural influences). Talker Echo disturbs the speaker who hears an attenuated and delayed echo of his/her voice. This is caused by a reflection on the distant end. EL-Echo Loss is defined at the ratio of the power (voltage) of the arriving voice signal to the power of the reflected echo signal expressed in dB (deciBels). If there is no echo, the loss is infinity. Listener Echo also influences the speaker who hears the signal from the other party followed by an attenuated echo of the signal. Listener Echo is caused both by a reflection close to the speaker and a reflection on the distant end. Loss of interaction is frustration by either party from the echo on the line.
If you want to know more, this information is part of OCS-101 and SIP Essentials 2.0c courses available onsite and online. Th online version is $299 for SIP 2.0c and $499 for OCS-101 Office Communications Server per person (volume and site license discounts available). For more information go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com.
Discounts are also available to members of the SIP Forum.
Related Tags: voice, noise, signal, suppression, energy, different
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TECHtionary.com Announces Podcast & iPod/iPhone Media Tutorials
April 18, 2008Multi-Media Speeds Sales Reduces Costs & Increases Customer Satisfaction
TECHtionary.com today announced new enhancements to its leading SIP/OCS courses. More than thirty new Flash (.swf), Quicktime (.mov, .mv4) and audio (.mp3) multi-media tutorials are now part of its SIP Essentials and Microsoft OCS-Office Communications courses. “These tutorials make advertising, sales training, marketing communications, technical and channel partner education available on the most-widely used devices in the world – Apple’s iPod/iPhone,” noted Tom Cross CEO TECHtionary.com. “It gives access to users who are time-constrained, mobility-driven and performance-pressured in today’s ruthless business world. Professionals can learn on the go and anytime they want to whether waiting at the airport or a customer’s office, driving down the street or riding on the bus/train/plane,” Cross added. Some of these topics are also available on podcasts from www.iTunes.com.
As a leading technical expert Matt Jolly said, “TECHtionary media animations provide a “pictionary” approach to many common support questions for customers who speak any language. This reduces the number and length of calls reducing provider’s expenses considerably. Next, it gives providers the ability to scale their business cost-effectively while maintaining even increasing customer satisfaction. In addition, TECHtionary animations provide the means to train agents and customers alike rapidly and easily. These been proven to reduce the delay in decision-making on the part of the customer, thereby accelerating revenues and reducing the customer sales cycle.”
Here is partial list of the multi-media tutorials available:
- Bandwidth & Packet Basics
- Internet & VoIP Introduction
- CODECs-COmpression-DECompression
- T-1 - ISDN
- VOIP 101 – Technical and Non-Technical Explanation
- SIP IM Presence
- TCP/IP & Firewalls
- ALG-Application Layer Gateways
- WiFi Roaming
- VLANs-Virtual Local Area Networks
- RTP versus RTCP
- RTCP-Real-Time Control Protocol-XR-eXtended Reports
- SIP “How It Works”
- SIP Basics
- SIP Event Notification
- SIP Trunk Replacement
- SIP Security Architectures
- SIP & QoS-Quality of Service
- SIP Applications & Future Outlook
- SIP & TCO-Total Cost of Ownership
- SIP Disaster Planning
- SIP Virtual Tie-Lines
- Integrated/Converged Access
- Key VoIP Options – Hosted/Managed
- SIP Security
- MPLS-Multi-Protocol Label Switching
These audio-video tutorials are available in number of media formats such as Adobe-Flash (.swf), .mp3 (audio only), iPod (.m4v) and Apple QuickTime (.mov).
All these tutorials are part of OCS-101 and SIP Essentials 2.0c available in the onsite and online courses. The online version is $299 for SIP 2.0c and for $499 as part of OCS-101 Office Communications Server online version per person or less with discounts. Discounts are also available to members of the SIP Forum. Join and support SIP Forum activities at www.sipforum.org.
Special Events Announcement:
All these tutorials are available as part of:
- CMP’s BCR SIP Essentials course offered May 12-13 in New York City. For more information go to: http://www.bcrtraining.com/course-info/sip.php.
- TMC University - One-day Introduction to Microsoft OCS - Tuesday, September 16, 2008
For more information and scheduling, please call Tom Cross at 303-594-1694 or cross@gocross.com.
Related Tags: tutorials available, multi media, media tutorials, tutorials, available, media
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One-day introduction to Microsoft OCS - Tuesday, September 16, 2008
April 17, 2008
Mark your calendars and signup for:
TMC University One-day introduction to Microsoft OCS - Tuesday, September 16, 2008
Related Tags: microsoft, Microsoft
TMC University One-day introduction to Microsoft OCS - Tuesday, September 16, 2008
What You'll Gain:
• Independent accreditation for completing the course.
• Independently certified evidence that you possess competencies in Microsoft OCS.
• Impressive certification from a respected source on your resume.
• Land lucrative consulting/ reseller opportunities.
• Immediately become the expert called upon to lead your company’s
Microsoft OCS strategy.
• Enhance your chances for a promotion
For more go here:
http://www.tmcnet.com/voip/conference/west-08/tmc-university-microsoft-ocs.htm
Related Tags: microsoft, Microsoft
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Interview with Marc Robins – Managing Director of the SIP* Forum
April 11, 2008
By Tom Cross
Tom Cross: Marc tell us more about the SIP Forum?
Marc Robins: The SIP Forum was originally founded 8 years ago as an industry association to promote the adoption of SIP by the IP telephony industry. Fast forward to today and now every product is SIP- compliant. Since the original “battle” has been won, now the SIP Forum is on to job #2 – solving interoperability problems to allow different SIP networks and various IP communications systems to seamlessly interoperate together. This is a critical need because SIP is a complex standard and there are a number of different options available to developers of services and equipment.
Tom Cross: Tell us about SIPconnect and the new SIPconnect Compliant Program?
Marc Robins: The SIP Forum is responsible for producing new industry technical recommendations that provide a specific set of rules and guidelines that determine which RFC and deployment option to use when and where. One example is the recent SIPconnect Technical Recommendation, which provides rules and guidelines for accomplishing SIP trunking, and in essence, direct IP peering between SIP-enabled IP-PBXs and SIP-enabled VoIP service providers. SIPconnect is a voluntary but peer-reviewed process
The SIPconnect Compliant Program is an associated certification program designed to certify equipment vendors and service providers as SIPconnect compliant and thus offer the highest degree of confidence that they will be able to accomplish trouble-free SIP trunking. To be approved in the SIPconnect Compliant Program, there is a process involving paying a fee (which can be applied to Full membership in the Forum) and completing a comprehensive survey. In addition, the applicant must demonstrate that they are interoperable through documentation examples including a self-assessment. There is a SIP Forum committee that reviews each survey – the SIPconnect Compliant committee essentially acts as a watchdog and makes recommendations for improvements with the goal of guiding companies along the right path. Nine companies currently hold SIPconnect Compliance.
Tom Cross: What else is the SIP Forum working on?
Marc Robins: The SIP Forum is now working on other technical recommendations on SIP Security and is examining other areas that need similar support such as UC-Unified Communications. There will also be other licensing programs as appropriate. The SIP Forum is a vendor and service provider-agnostic, non-profit association. The SIP Forum relies on volunteers to help drive the cause, the work and the new initiatives. There are currently 40 Full Member companies that support the SIP Forum, For a full list of Full Member companies, visit www.sipforum.org. If you have any questions regarding the SIP Forum, please call Marc Robins, SIP Forum Managing Director at (718) 548-7245 or email marc.robins@sipforum.org
----------------
*Definition and brief explanation - SIP-Session Initiation Protocol is a "signaling" system for connecting, monitoring and disconnecting connections across the internet using intelligent devices (end points). SIP is not about voice as it is about “presence” and “rich media communications” providing enhanced information (e.g. maps, availability), intelligence (e.g. automatic event-notification), and context (e.g. reference material) to single (1-1) and multi-party (1-X, X-1, X-X) communications sessions.
The benefits of SIP briefly is that with SIP Trunking, the IP media stream coming from within the enterprise stays as an IP media stream and passes to anywhere within the enterprise or across the boundary of the enterprise to another enterprise via IP. This reduces the need for local telephone systems using instead hardware media gateways at the enterprise edge and carrier edge (often referred to as the PSTN) producing considerable savings. In addition, considerable savings can also be found in eliminating expensive telephone desksets by using intelligent softphones.
If you want to know more, register for SIP Essentials Public Sessions in New York – May 12-13, Specific course details can be found at: http://www.bcrtraining.com/course-info/sip.php or this information is also part of OCS-101 and SIP Essentials 2.0c available in the onsite and online courses. The online version is $299 for SIP 2.0c and for $499 as part of OCS-101 Office Communications Server online version per person or less with discounts. For more information go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com Discounts are also available to members of the SIP Forum for online and custom courses.
Related Tags: sipconnect compliant, compliant program, Forum, forum, SIPconnect, sipconnect
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Session Border Controllers & Media Gateways Soup Up SIP Communications
April 10, 2008MG’s & SBC’s Dramatically Change the Enterprise Communications Landscape
The following pictutorials are examples of customer applications of SBC-Session Border Controllers with an emphasis on MG-Media Gateways:
- SBC-Session Border Controllers connect SIP networks
- MG-Media Gateways Next-Generation Networks
- Customer Applications of Media Gateways
A few basics - SBC-Session Border Controllers connect SIP devices to the PSTN and MG-
Media Gateways also connect SIP devices to the PSTN. The difference is SBC's are provider-class systems, whereas MG's are designed for local connection to the PSTN.
The primary function of the SBC is to serve, basically, as a SIP-aware NATing Firewall. SC-Session Controllers or SBC-Session Border Controllers are access devices operate at Layer 5 Session Layer, where as routers operate at Layer 3 Network. Some of the key SBC-SC functions are:
- Secure network peering - private and public to enhance performance
- Topology hiding - using various types of inter-AS-Autonomous System features as well as separating media (voice) and hide signaling (IP addresses) data streams (traffic)
- Border call routing - routing at AS level rather than with interior protocols
- Interoperability - access/restrict to reduce voice spam
- QoS & Call Admission Control - load/jitter correction
- Billing systems interoperability - reduce billing errors
- NAT-Network Address Translation - routing for maximum performance
- DNS-Domain Name Service and ENUM (E.164) support
- CALEA-Communications Assistance for Law Enforcement Act
- E911 support
- Compatibility with customer billing
- Dialect conversion
- Protocol conversion
- Codec conversion
- Firewall restrictions
- Wholesale and Transit peering - various types of stateful (interconnect different networks such as H.323, MGCP-Media Gateway Control Protocol and SIP-Session Initiation Protocol) and stateless (same networks) Session Controllers exist depending on the VoIP Network features required.
- Other features, check with manufacturer.
In other words, even a high-volume company might only need a few MG's or SBC's to provide carrier-class performance including a vast array of features.
In the pictutorial are some examples of ON/OFFnet SIP calling via MG & SBC
- ONnet SIP Calls & ONnet Conference Calls.
- Key point: no need for MG or PSTN with SIP network peering even fewer GW needed.
- OFFnet SIP Call Flows
- Key point: No need for local PBX/key systems or trunks with the need for only a few gateways nationally or globally.
- OFFnet PSTN calls via MG-Media Gateways
- Automatic MG call re-routing in case of failure to backup MG
- Automatic call re-routing in case of failure via SBC
Shown in the pictutorial are some examples of Common SIP Calls:
- Call Signaling via Proxy server
- Media Transmission Directly
- Parallel Forking Simultaneous Ring
- Sequential Forking - Sequential Ring - "Call Office, Call Home, Go to Voice Mail"
- Call Authentication via Proxy server
- Redirect Call Forwarding via Proxy server – UAB, then Call Processing - via Proxy - UAC
Media Gateways
That is, MG's would be strategically located across the continental or world and interface the PSTN only as necessary.
Shown in the pictutorial are some examples of customer applications of MG-Media Gateways:
- Connection of IP-PBX to PSTN
- Connection of IP-PBX to PSTN & SIP trunk provider
- Survivable connection to SIP trunk provider
- Connection of PBX & IP-PBX to PSTN & SIP trunk provider
Hairpin or Hairpinning - calls come in from PSTN and routed back out to PSTN
- Connection of IP-PBX to Hosted VoIP provider
- Connection of IP-PBX & PSTN to Microsoft OCS Server
- Multiple SIP-OCS proxies - SIP to SIP calls, TDM to TDM calls and SIP to TDM calls
If you want to know more, this information is also part of OCS-101 and SIP Essentials 2.0c available in the onsite and online courses. The online version is $299 for SIP 2.0c and for $499 as part of OCS-101 Office Communications Server online version per person or less with discounts. For more information go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com Discounts are also available to members of the SIP Forum.
Related Tags: media gateways, border controllers, session border, pictutorial examples, customer applications, media
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New SIP Essentials Public Sessions in New York – May 12-13
April 9, 2008“Quintessential Guide to SIP:” Two-Day Intensive Course
SIP Essentials is produced by TECHtionary.com, the world’s largest multi-media knowledge library. “This indepth and critical course on SIP combines the resources of BCR and the multi-media tutorials and research of TECHtionary,” noted Thomas Cross – CEO TECHtionary. “After more than a year of research and exhaustive interviews with users, a dealer, providers and industry experts, SIP Essentials is the first major course to address current and future applications.” According to Paul C. Daubitz, President of ATI-Telemanagement (http://www.ati-telemgt.com/), “SIP is the next major significant development for the integration of voice and data. Understanding the critical technical and business issues is crucial for implementation, use and applications development. TECHtionary’s SIP Essentials is the quintessential look at SIP, SIP Trunking, business applications, Security, QoS and more. Scanning the literature
Here are just some of the key tutorials in SIP Essentials:
- SIP Basics
- SIP Trunking
- SIP QoS
- SIP Firewalls and Security
- SIP Applications
- SIP TCO-Total Cost of Ownership
- Integrated/Converged Access
- Key VoIP Options – IAS, Hosted, Managed
- SIP Total Tutorial with Future Outlook including a look at Microsoft’s OCS-Office Communications Server
Specific course details can be found at: http://www.bcrtraining.com/course-info/sip.php. , for registration call 800-227-1234
This course is also available for private onsite and online delivery. For licensing, customizing and delivery opportunities, email cross@gocross.com or call Tom Cross @ 303-594-1694.
Related Tags: course, essentials, Essentials, TECHtionary, techtionary, applications
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OCS Exposed – SNAT/DNAT – Load Balancing
April 6, 2008The load balancer provides scalability and availability across multiple servers that are connected to a centralized database on the OCS Back-End Database. Only one load balancer is required, there can be two logical load balancers—one for the Front End Servers and one for the Web Components Server. Two are recommended for installations using the Enterprise Edition-Expanded configuration. In OCS, SNAT-Source Network Address translation load balancing is recommended for ease of deployment. Each SNAT IP address on the load balancer limits the maximum number of simultaneous connections to 65,000. DNAT-Destination Network Address translation can also be used for load balancing for the Enterprise user pool.
In the second slide – the Table above lists the Port addresses used to configure the LB-Load Balancer. A load balancer is required in an Enterprise pool that has more than one Enterprise Edition Server. Two VIP-Virtual IP addresses are needed for either two logical or separate physical load balancers. When you configure the load balancer, check with the firewall and DNS administrator for at least one VIP-Virtual IP address and FQDN-Fully-Qualified Domain Name for the load balancer (one for each logical load balancer), as well as a static IP address for every server in the Enterprise device pool.
If you want to know more, this information is also part of OCS-101 and SIP Essentials 2.0c available in the onsite and online courses. The online version is $299 for SIP 2.0c and for $499 as part of OCS-101 Office Communications Server online version per person or less with discounts. For more information go to http://www.techtionary.com or please call Tom Cross at 303-594-1694 or cross@gocross.com Discounts are also available to members of the SIP Forum.
Related Tags: balancer, enterprise, address, Enterprise, server, online