Saturday, September 24, 2011

The Benefits An SIP Trunking Provider Can Bring To Your Business

In case if you are wondering a Session Initiation Protocol (SIP) is a method of communication that allows businesses to make and receive a call over the internet, you should continue to read on. The SIP trunking provider is the service that provides the service to business users. For organizations that already have the internet and receive their phone lines from the same source, it is a simple task to switch to SIP trunking.
Switching to SIP trunking can provide organizations with several benefits and there are a number of providers to choose from. Not least of which, is the improved clarity of the connection on phone calls, compared to the old-fashioned phone systems. Providing you have a good internet service connection, the quality of your telephone calls will be greatly improved, cutting out all the unwanted interference.
One of the main benefits that users have found by switching to this method of communication is a reduction in the cost of their telephone line rental and in the cost of their calls. This can be a huge benefit for those who rely on their telephones to contact suppliers and customers, particularly if they are based overseas.
The ability to use conference and video calling and to hold meeting with individuals in various locations saves time and money. There is no longer the need to juggle conflicting schedules to ensure everyone is in the one venue at a specific time. As long as all the parties have access to the internet, there is no need to pay for trips that can be a major business expense. It is often better for business, to be able to see the person you are talking to, rather than just hear them.
There is not a need to go through what can be a costly exercise to change all your contact phone numbers on all the preprinted paperwork, letter heads, business cards and advertising copy. You do not have to notify all your customers and suppliers of a change of a phone number. Your provider can arrange for your business numbers to remain the same.
Some businesses find when they need to expand the number of telephone lines they use, that they are restricted. SIP trunking makes it much easier to add more telephone numbers for your business. It can be done almost instantaneously, rather than having to wait on physical changes being made to the phone lines and systems.
SIP providers can offer features such as Advanced Call Management. This service has benefits for your business and your callers. Callers are advised they have been placed in a queue, and calls are processed in the order they come in. Calls can be redirected when no-one is available to answer on a specific number. With the auto-responder, callers are never left wondering if they have been cut off.
There are several SIP trunking providers available in the market place. They all have their own unique packages to offer customers. It is worthwhile shopping around to find out what is available as this looks like being the way forward in communications.
Finding where to get SIP Trunking Provider may not be that hard when you know where to look for them. If you are looking for deals and advices on SIP Trunking Provider take a look at the information available to find out more.


Article Source: http://EzineArticles.com/6538969

Wednesday, August 17, 2011

SIP Trunk Features and special functions

  • A Session Initiation Protocol (SIP) network is a type of Voice Over Internet Protocol (VoIP) service designed for businesses with a private branch exchange (PBX) phone system. A SIP Trunk lets the company continue to use its connected telephone system, but, instead of using a digital or analog telephone line, you communicate with others via a data connection like the Internet. These systems are generally cheaper than standard PBX systems, but just as scalable.

Calling Features

  • Like a standard telephone line, you can make outbound calls and receive inbound calls as you normally would, including calling 800, 888 or 877 numbers. Some countries cannot call toll-free numbers in the United States, which could hurt your business presence. You can create local numbers wherever you have offices, eliminating the need for toll-free numbers. For example, if your headquarters is in New York, but you have an office in Los Angeles, you can get a Los Angeles area code for that office while still using a single SIP Trunk service. Generally, you can make long distance or international calls for a set or discounted rate with SIP Trunk systems.

Expense

  • While you will have to pay for a few items up-front, businesses save money in the long run since they are only paying for the service for both your voice and data communications. This means that your SIP Trunk provider rolls your Internet and telephone services into one package, and you usually start seeing savings immediately. Some of the up-front costs include investing in an Internet Protocol (IP) PBX, firewalls for your SIP Trunk, IP phones and soft clients, which you will usually use for your mobile devices. Once your system is set up, you should see a return on your investment in savings within six months.

Other Features

  • Unlike traditional phone systems, SIP Trunk systems are not connected to a single phone line, meaning that your SIP system can route calls anywhere in the world. This means that if your New York office is closed for the day, you can have your calls routed to the Los Angeles, ensuring that your clients reach a live person more often. If you have international offices, this system can route the calls to those offices as well. These systems can handle any type of voice or data communication, including fax, text messages and video files. SIP Trunk systems also don't take very long to set up, usually only a week or two, and SIP Trunk providers generally offer disaster recovery, which allows you to retrieve your documents should your system crash.

Sunday, July 17, 2011

Getting Networking Jobs

There are many factors that go into choosing a job, however if you have chosen to apply for a networking jobs then you have to know a few things about how to answer networking interview questions.
The fact is that if you have applied for networking jobs then you know what you're talking about when it comes to networking, you should know the ins and outs of IPv6 connectivity, SQL, SIP Trunking and VPN. You will most likely use a Linux based operating system, just because you can.
However knowing your stuff when it comes to the inner workings of networking may look good when it is written down on your CV, but in an interview it will not get you the job, you may be the most experienced or qualified networking guru in the entire world but if you can't express that to the person interviewing you then you don't stand a chance.
You must understand that networking jobs are highly sought after (especially in countries such as India and South Africa) and the level of competition is high. And the key to being chosen could be a simple thing such as preparation. Typing in the right words into a search engine and taking an hour to read a website detailing networking interview questions and answers will definitely put you head and shoulders above the competition.
The thing with networking jobs is that the same networking interview questions will be asked at every interview, this is because although the industry around it changes very quickly, the bare bones of networking jobs stays pretty much the same, with the exception of some large changes that happen every so often (which any networking employee or prospective employee will be aware of month/ years n advance) the job stays very familiar, things such as TCP/IP and L2TP are the industry standard and always will be.
The thing to remember is that as long as you have the knowledge and are prepared for the interview then you should stand a great chance at being successful at your interview. Read up on current events and make sure that you have some basic understanding of how to use you body language and how to answer specific networking and behavioral questions then you shouldn't need to worry about anything except for enjoying your new job.
Make sure that the questions that you prepare for contain some current events, such as the recent world IPv6 day, that was tested by almost every respectable network operator and website in the world.
Dan Ray is an expert in the a networking interview questions field and can help you beat the competition when applying for networking jobs.

Wednesday, June 22, 2011

Patton Has Added Three New VoIP Gateway and SIP Trunking Solutions to Its Portfolio

Patton Electronics, a VoIP solutions provider, has recently added to company’s wide product line three new SmartNode PRI-class VoIP products: the SN4950 Gateway-Router, the SN4940 Gateway and the improved SN4960 Enterprise Session Border Router (ESBR).
These three IBM-certified solutions were specifically designed to empower the use of SIP-trunking and media-gateway services.
According to company’s officials, the company has broaden its SmartNode product line with a purpose to assure that enterprises, carriers and integrators have the possibility to select a solution that better fits network infrastructure requirements, business processes and communications strategies.

Grandstream HandyTone HT286 VoIP Phone Adapter
Cisco PAP2T Internet Phone Adapter with 2 Ports for Voice-over-IP
Bugaboo - Peg Perego Car Seat Adapter

Sunday, June 19, 2011

How to Explain SIP to a Non-Technical Person

SIP... or Session Internet Protocol... isn't necessarily a new communications concept. It has been around awhile although it seems to be garnering a resurgence in Telephony applications today. But... how do you explain what SIP is when asked?

Good luck with this one, trying to explain SIP to technical people is tough enough.

Here's some very simple short statements that may help...

- SIP enables telephony over the internet network

- SIP allows us to packetize and prioritize voice traffic over digital circuits.

- SIP is a way voice is packed into a digital signal that is then enabled for transfer through the internet.

- SIP digs a channel in an IP network so voice/video can flow between two (or more) places. When you finish talking, SIP shuts the channel up.

- It's an internet protocol like HTTP for web browsing, only this one is used to make a phone-like connection between computers, pda, voip-phones or other devices that can talk over the internet.

- SIP is a protocol that allows unlike mediums to communicate. All you really need to know is that SIP is the new PRI and is more cost effective from a trunking perspective.

- SIP has nothing to do with the internet.... regardless of where, when, or how voice traffic is being transmitted. f it's being sent as 0's and 1's... SIP is what differentiates voice from all other data.

- SIP enables you to eliminate the cost of maintaining two networks (POTS + Ethernet) by putting your phone traffic on your Ethernet network.

- SIP is a business-class, integrated voice and data service with connectivity provided to your IP-PBX (a telephone switch that supports voice over IP)

Or... you may explain to a non-technical person by describing the SIP VoIP operation like this:

1. Callers and callees are identified by SIP addresses.

2. When making a SIP call, a caller first locates the appropriate server and then sends a SIP request. (The most common SIP operation is the invitation).

3. SIP or VoIP is a technology that allows you to make calls between devices, be it over the local network or over the Internet (Managed or un-managed). SIP is a standards based technology that behaves very much like your old telephone line but just uses the Internet as its medium.

4. Instead of directly reaching the intended callee, a SIP request may be redirected or may trigger a chain of new SIP requests by proxies.

5. Users can register their location(s) with SIP servers.

6. SIP messages can be transmitted either over TCP or UDP

7. SIP messages are text based and use the ISO 10646 character set in UTF-8 encoding.

8. Lines must be terminated with CRLF.

9. Much of the message syntax and header field are similar to HTTP.

10. Messages can be request messages or response messages.

Now you are armed with some basic background in simple (or as simple as possible) terms.... to explain what SIP is and does. Should you need additional help of a more technical nature in deciding what SIP solution would work best for your given business application.... take advantage of the resources available at Broadband Nation.

Michael is the owner of FreedomFire Communications including Business VoIP Solution. Michael also authors Broadband Nation where you're always welcome to drop in and catch up on the latest BroadBand news, tips, insights, and ramblings for the masses

SIP Trunking
SIP: Understanding the Session Initiation Protocol (Artech House Telecommunications)
SIP Demystified

Sunday, May 22, 2011

Relationship Between Business and VoIP Services

VoIP telephony service has undoubtedly come as an upgraded mode of communication and has gradually begun to change the entire communication structure of businesses and even residentials. It offers clear advantages over the traditional public switch telephony network or PBX systems. Especially for corporates, VoIP has come as a breath of fresh air and provides them with a single solution for all their communication needs.

The major benefit of VoIP telephony services is its cost-effectiveness. The phone bills are lessened with VoIP. In fact, you end up saving almost 50% on every single call you make. The yearly savings on the telephone bills thus decreases substantially.

Best business VoIP providers also offer a number of services to corporates which help them in improving their functionality. For instance, some renowned service providers help in interconnecting the different branches of the organisation, thereby unifying their communication structure and helping them coordinate their work, at no extra cost.

Such services are either not provided by traditional modes of communication or else businesses are charged heavily for it. Moreover, voice over IP comes along with the mobility feature which makes it ideal to be used by businesses, either by businessmen who travel a lot or work from home. VoIP services help the user to get all the calls on the same number and that too at half the cost of PSTN.

Voice over internet protocol telephony services offer the option of using a DID number, which enables the user to get the calls on a number which is not based on that particular geographical location. That is, you would be able to receive a call from a client in the U.K when the number which is being dialed is that of U.S.A. This is especially useful for contact centers that provide customer support from other countries.

You can compare VoIP provider on the basis of the quality of service offered by them. The parameter that could be followed for judging the quality can be based broadly on 2 factors. Firstly, look for an established VoIP service provider with multiple switches and point of presence. This provides not just reliability but multiple POP's and switches, which means that the call would be transmitted via the best and shortest route possible, thereby reducing the latency to bare minimum.

Secondly, always ask for the ACD (average call duration), ASR (average success ratio) and PDD (post duration delay) ratios from your VoIP provider. An ACR of 4-5 minutes is considered okay, whereas anything above 6 minutes is excellent. ASR should be at least 40-50%, and anything above 60% would provide excellent quality service. PDD should be not more than 4-5 seconds, and anything below 2 is considered excellent.






Polycom VVX 1500 - IP video phone - SIP - 6 lines
Brand New SIP IP DECT CORDLESS PHONE by Panasonic Warranty
ClearOne Max IP - Conference VoIP phone - SIP




Monday, April 11, 2011

How to Use the Data Card in a NEC PBX

The NEC NEAX 2400 PBX system is designed to be part of both simple and complex phone system configurations. The PBX system is designed with four internal data or I/O card expansion slots for use in expanding its system capabilities. Data or I/O cards give the NEC PBX system the ability to handle more phones in the system, and phone extensions per phone. Once a data card is installed into the NEC PBX, it is automatically available for use by the system.


Instructions

1) Unscrew the six screws on the back panel of the NEC PBX to remove the panel. Once the screws are removed, lift the back panel upwards and off.

2) Insert the data card into one of the four empty data card slots on the center left-side of the PBX's main board, aligning the metal contacts on the data card with the metal contacts in the card slot. Push the card in gently until it snaps into place.

3) Replace the back panel onto the unit, sliding it downward into place. Then, replace and tighten the back panel screws

Polycom SoundPoint IP 331 - VoIP phone - SIP - 2 lines

Grandstream GXV-3000 - IP video phone - SIP, SIP v2 - 3 lines

Aastra 6757i CT RP - Wireless VoIP phone base station w/ corded handset - WDCT - SIP - 9 lines + 1 additional handset(s)

Wednesday, April 6, 2011

SIP Trunking - The Key Questions Answered

Q: What is SIP Trunking?This is a way to link your PBX to a phone network with the help of a data network as well as the protocol that is called SIP which stands for Session Initiation Protocol.

Q: What are the Benefits? You are provided the opportunity of making calls at a cheaper rate, with cheaper infrastructure and you will not be integrated with any telecoms circuit.

Q: What's the call quality like? Once SIP trunking is properly set up the quality of the call is as good as or even better than your regular phone service. On the other hand if it is not set up by an expert you will have packet loss and latency in your connection which will affect your call quality negatively.

Q: I'm a consumer. Could I benefit? If you have a standard internet connection as a consumer you will not get a good quality call signal and so the reduction in cost won't be to your advantage. If you don't mind having fluctuations in your call quality though you can use this method but ensure you have a back up if it fails miserably.

Q: Can SIP Trunking deliver a business-class service that's as reliable as my existing setup? In order to get great quality, you will need to get a minimum latency, low jitter and minimal packet loss network to connect your PBX to the provider of your SIP trunking services. Your best bet to ensure good call quality will be an ISDN30 circuit along with an internet connection that is dedicated solely for this purpose. This way the amount you save will be an advantage as you will have clear call quality.

Q: Will SIP Trunking be compatible with my existing PBX? Once the PBX can understand the signals of your SIP trunk you should have no problems.

Q: How should I choose which provider to use? By looking for ones that offer business class services, continuous telephone support (24/7) and multiple telecoms carriers.

Q: Is SIP Trunking a fad? This is a definite no. Many big companies use SIP support for their products and services. These companies include Cisco, Verizon, Microsoft, Mitel, Siemens, Avaya, Panasonic, Nortel and Samsung. These are just a few as increasingly more and more businesses are replacing their ISDN's with SIP.

Q: How much can my business save by switching? There are a number of factors that affect the answer to this question. Some are the quantity of calls made, when the calls are made, which numbers are being called and how long the calls last. Other factors include the minimum call charge as well as the rental costs of the ISDN circuits and the length of the contract.
Your best bet for finding out how much you could save is to find a SIP trunk provider and send them your most recent phone bill. They will use this to provide you with an estimate of how much you could save for your particular business.

Q: How long does it take to switch to SIP Trunking? It can take as long as it will take you to get an internet connection of 2mbps for less than 30 channels and for in excess of 30 channels it will take as long as it takes for you to get a 10mbps internet connection.





Tuesday, March 29, 2011

Super Charging the SIP Trunk

I was on a panel about Supercharging the SIP Trunk Sales by Broadvox. It sounded similar to the panel I did at CVX West 2010 on Upselling the SIP Trunk sale.
This one had to go a little more basic to What is SIP? SIP Trunking is how carriers power dial-tone to an IP-PBX. Beyond the definition of the term, there is the concept that this specification for a voice packet to provide dial-tone is the foundation of the Next-Gen Communication platform. Start with the SIP trunk as the basic circuit needed for communications. Then other applications that utilize SIP sessions can be integrated into the platform. (In many cases, the platform is the PBX or the softswitch.)
One point that I made: All voice will soon be Voice over IP. Cellular is. Long Distance is. Dynamic T1 is. Cableco digital voice is. AT&T is turning off the PSTN with the approval of the FCC in less than 10 years. SIP has become the de facto protocol for Voice over IP - beating H.323, MGCP and others. Agents (and customers) have to get on-board with this fact. SIP is here to stay.
Yes there are issues. Most notably Fax-over-IP (which many companies are working on, T.38 not withstanding); HD Voice; alarms, elevators, credit card processing machines; and inter-operability between the SIP provider and the gear attached to the Trunk. Since SIP is a specification of about 30 RFC's, it is interpreted differently by manufacturers and providers. Hence, inter-operability is extremely important. It is not like a PRI, which is a standard, with just 2 configurations available in any class 4 or 5 switch or any PBX system. That makes inter-op easy. Today. Maybe not so much in 1988.
Our panel then went a little crazy talking about Unified Communication and all the possible UC components that could be mounted on that SIP trunk - like SMS/text, IM/chat, Video, Presence, conferencing, ACD, IVR, etc. Unfortunately, it's not that easy and most of the integration is with the gear (PBX, IAD, softswitch, SBC).
During Q&A, some asked how to explain SIP to a customer. I said, Don't. Would you explain how the engine works in a car? Focus on the benefits, the reliability, the way it will help the business and you won't have to worry about saying VoIP or SIP.
The panel had to explain that SIP trunks mean different things to different carriers. In some cases, a SIP trunk is just a call path. In other cases, a SIP Trunk is a circuit containing many call paths. It's confusing for the agent and the customer.
There are two ways to deliver SIP Trunking: over an IP circuit and over the Internet. Big difference. SIP sessions over the Internet lack quality of service. Jitter and latency will affect the voice quality. SIP Trunks that ride private line, virtual circuit or MPLS circuit have not only the best call quality but enhanced security.
Many ITSP's started out as ISP's (or at least with an ISP infrastructure). At the NOC, the ITSP has aggregation circuits for DSL, MPLS, private line from ILEC's and CLEC's. Some even have aggregation with a cableco. This means that the ITSP - in its network footprint - is delivering VoIP service to you on a private network with QOS of some level. The customer is receiving Internet over the same circuit but the egress to The Web is AFTER the aggregation point and separate from the softswitch.
Major MSO's deliver their Enterprise VoIP product as a physically separate VLAN. That offers QOS and security as well.
At XO,  Pete Davis and I have collaborated on a concept called The UC Sandbox. XO has all the components for UC available for sale. Hosted MS Exchange, Blackberry server, Hosting, Collocation, IM/chat, Hosted PBX, SIP Trunking, Contact center application, web & audio conferencing, video conferencing, data storage, Anywhere service, and more can be provisioned for the customer as a single individual service or as a bundle of components. The bonus for VAR's is that if they have a business selling email, security, what-have-you, they can mix-and-match components from XO or from a variety of providers to put the best solution together for the customer. That is the true value of VoIP, SIP and the future of Cloud.

SIP Trunking

SIP: Understanding the Session Initiation Protocol (Artech House Telecommunications)

Western European SIP Trunking Market, 2007-2012

Wednesday, March 23, 2011

How to Edit a Sip Trunking

Voice over Internet Phone service providers use Session Initiation Protocol, or SIP, to provide customers with access to the private broadband telephone network. SIP includes server information as well as address, username and password and other information needed to access VoIP networks. SIP information stored in hardware or software IP phones ensures that calls route through the correct network before connecting to external VoIP phones or landline telephone systems. If you switch VoIP services or providers, you must edit the SIP settings before you can use your phones to place or receive calls on the new network.

Instructions

Reconfigure Software SIP Phones

1) Launch the software VoIP phone application on your computer.

2) Click the "File" or "Tools" menu or tool bar link. Click the "Preferences" or "Settings" menu option to enter the software phone configuration or settings utility.

3) Enter the new username and password assigned to you by the new VoIP telephone service provider. Replace the entry in the "External Server," "VoIP Server IP Address" or "SIP Server" field with the public IP address of the new provider's SIP site IP address. Save the configuration settings and exit the application.

4) Restart the software phone program and wait for the application to connect and authenticate with your new VoIP provider's SIP server. After the software establishes the connection, use the software VoIP phone to place and receive calls via the provider's server on the Internet.

Change SIP Settings on a Hardware IP Phone

1) Unplug the AC power adapter from the hardware IP phone and leave it disconnected for about 30 seconds. Reconnect the AC adapter and power on the IP telephone. Wait for the hardware IP phone to initialize and for the configuration IP address to appear on the phone's display screen. The IP address should be similar to "192.168.1.101."

2) Go to a computer connected to the same router as the hardware IP phone. Open a web browser application and browse to the IP address displayed on the phone. Enter the username and password and log in to the IP phone's web-based configuration utility.

3) Click the "IP Settings," "SIP Settings" or "VoIP Account Settings" link in the configuration utility. Replace the entries for the VoIP account username and password, SIP server name and SIP server IP address with those provided to you by the new VoIP service provider. Click "OK" or "Update."

4) Wait while the configuration utility downloads the account changes to the hardware IP phone. Reboot the IP phone when prompted to do so.

5) Wait for the IP phone to boot and connect to the new VoIP network. Use the hardware IP phone to place and receive calls using the new VoIP provider's network.

Internet Communications Using SIP: Delivering VoIP and Multimedia Services with Session Initiation 


Western European SIP Trunking Market, 2007-2012 


U.S. SIP Trunking 2008-2012 Forecast: SIP Trunking Wins This Battle, But Who Wins the War? 

Sunday, February 27, 2011

VOIP,SIP,Gateways,Gatekeepers and Codecs (Take Me Away Please)

If you do anything in the VoIP world you hear terms like gateways and gatekeepers, SIP, SIP trunking and Codecs well here a few quick definition of each.

A VoIP gatekeeper (not to be confused with gateway) is an optional feature for a VoIP network, though many might argue that. A gatekeeper is used for routing and central management of all endpoints in a given zone.
This includes the management of terminals, gateways and MCU's (multi point control units).

The gatekeeper provides logic variables for proxies and/or gateways in a call path to provide connectivity with the Public Switched Telephone Network (PSTN), and to improve Quality Of Service (QoS), and to enforce security policies, which can be very important depending on the type of call and who you are trying to speak too.

A VoIP gatekeeper also provides address translation, bandwidth control, and access control to a network of VoIP terminals and gateways. This grouping of elements (gateways, gatekeepers, VoIP terminals) under control of a gatekeeper is defined as an H.323 Zone.

A VoIP gateway may also be known as a Media Gateway, Soft-Switch, Media Gateway Controller, SIP Server, or other device that handles VoIP data and signaling traffic.

A gateway is a network point that acts as an entrance to another network, like a traffic cop. Now on the Internet, a node or stopping point can be either a gateway node or a host (end-point). Both your home computer and the computers that serve pages to users are host nodes.

The computers that control traffic within your company's network or at your local Internet service provider (ISP) are gateway nodes.

SIP is the Session Initiation Protocol, which is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences. SIP was actually created to replace the H.323 protocol.

Now you also have SIP trunking which unlike in traditional telephony, where bundles of physical wires were once delivered from the service provider to a business, a SIP trunk allows a company to replace these
 traditional fixed PSTN lines with PSTN connectivity via a SIP trunking service provider on the Internet.

There are three elements necessary to successfully deploy SIP trunks: a PBX with a SIP-enabled trunk side, an enterprise edge device understanding SIP and an Internet telephony or SIP trunking service provider.

And finally we have Codecs, a Codec" is a technical name for "compression/decompress or compressor/decompresser" and "code/decode". In the VOIP world we deal with 2 Codecs for the most part G.711 and G.729.

To make this even more confusing in North America we use mu-law G.711 and in Japan they use a-law. G.711 is 64Kbps and with overhead about 90Kbps and G.729 is 8Kbps and with overhead about 32Kbps.

I know the VOIP world can be confusing and of course we love our acronyms, like no other industry on the planet but at the end of the day voice is still voice. When a user picks up the phone they want dial tone and regardless if they have to dial a 9 to get out or 4 digits to get Joe in accounting they want it to work.

Monday, February 21, 2011

SIP: how Interdependence can boost revenue during slow economy

Many solution providers like vars seem to struggle to keep up their revenue as their clients have less money to spend. This probably also goes for other types of business to business companies. What I hear often from
 vars is that their customers are postponing the so needed upgrade of their outdated phone system for example.

As revenue from the existing client base is going down, I see two obvious options:
Grow your client base
Find an alternative way to present your solution so that it makes financial sense to the current customer base


The first point, finding new clients, sounds pretty obvious but may be hard to accomplish as your fellow solution providers are trying to do the same. It is still very much worth the effort but you have to really come up with a unique selling point to distinct yourself from the competition.

The second point may seem even harder at first. When the capital expense of a new PBX phone system for example does not fit their budget, make it fit. Perhaps the customer is using an old technology and by presenting a phone system that can save them money on their telecom usage expense, it can suddenly start to make sense to upgrade. For example what if you present a VoIP based solution as SIP Trunking, if they had analogue phone lines until now this may be a significant cost saver. If they don't have the capital, it may even still make financial sense to lease.

What does this have to do with Interdependence versus Independence?

It does not necessarily have anything to do with it but it may make things easier. Independence may be something that many business owners are actively perusing: becoming as independent as possible so that they have total control over their own business. They may fail to see the mutual benefit of interdependence.

Patrick Oborn, co-founder of Telarus, has written a great article about interdependence versus independence in regards to master agents. As we work with a lot of var partners, we found this to be an inspirational article and decided to show what interdependence can do for our var partners.

When a solution provider partners with a master agent, the relationship can only be truly successful if it is of mutual benefit. As with any relationship it is a two way street.

Thus comes Interdependence, growing your business by growing each other's business.

The relationship we are aiming for with our var partners is one of interdependense. We as carrier services consultants come across customers that have just ordered a new PRI circuit and they need a new PBX. Guess what, our var partner gets the opportunity. And that goes the other way as well: when our var partner has a client in need of a new T1 line, or any other carrier circuit - guess who he refers them to?

Besides sharing and referring leads between the interdependent companies (in this case the solution provider var and the carrier services consultant), the involved parties also grow their value proposition to the customer by offering a more complete solution. By partnering with a master agent, the var partner is able to offer carrier services without the hassle of dealing with all the carriers. The var partner grows their revenue this way, and he may be able to justify the new equipment purchase for his customer by comparing overall cost. The carrier consultant can even help by offering a circuit with a vendor that will amortize the equipment cost, so that there is less capital expense. And we as carrier services consultants are happy as well because this brings us more revenue.

True Interdependence is vital for a successful business partnership, don't shy away from it.

Tuesday, February 15, 2011

sip trunking solve problem

Problems with your SIP trunk, be they registration, audio or other failures, could be due to an issue with either your ISP or your enterprise network and equipment. Most SIP trunk problems could be isolated to one of the other of these sources by call flow mapping and conducting simple testing at the VoIP demarcation point.

    Registration Problems

  1. If your registration is unstable, check the stability of your internet connection, then try overriding refresh time in the trunk's overflow and failover rate settings to around 20 seconds. If you can't register at all, check that you're using the right password and authentication, ensure your ISP supports SIP by registering a soft phone and examine traffic before and after your firewall to see if it is filtering SIP traffic.
  2. Audio Problems

  3. If you experience poor audio quality, check and reset network equipment, then ping a location nearby for an extended length of time and observe any jitter or response delay. Make sure that your ISP and every component between the internet and the PBX respects the QoS and that you're not forcing any transcoding. If you're only getting one-way audio, check if your router or firewall is misinterpreting audio packets as an attack.
  4. Other Failures

  5. Other SIP trunk failures could be due to interoperability issues between your SIP trunk provider and IP PBX or applications and legacy services, pointing to a need for session border controllers, gateways or software upgrades. If all your SIP trunks have been deployed through a central architecture, try instead deploying an equal number of SIP trunks at all PSTN locations.
SIP: Understanding the Session Initiation Protocol (Artech House Telecommunications)

Sunday, January 30, 2011

SIP Trunk - What You Must Consider

If you are looking for ways to improve the telephone system of your company, then perhaps you have heard recommendations from friends and family about getting a SIP trunk service. This service connects your company's own telephone system (called PBX or Private Branch Exchange) to the internet via a SIP soft switch, instead of following the traditional connection to the public telephone system (called PSTN or Public Switched Telephone Network).

So when you dial a number to make an outgoing call, the softswitch will analyze whether you are trying to reach a PSTN phone number or a SIP-enables system as well. After that, the softswitch will then route your call to wherever it is directed.

But this is when the difference takes place, because the soft switch will not pass your phone call to simply travel by means of normal telephone lines. It will go through the internet. This is how the SIP trunk service basically works. So, the question for you now, why bother getting this service when it does the same function as a normal telephone system?

As the SIP trunk service works over the internet, is measures the presence of telephone calls in terms of bandwidth. And because of this, unlike typical phone systems, it can support multiple phone calls at a single given time as long as it can handle the bandwidth capacity.

Wouldn't you want this advantage for your company? Also, SIP trunking removes the complicated abundance of hardware and wirings just to support the flow of calls, because the only thing needed is to use the servers of the chosen provider.

Getting a SIP trunk service is not as simple as getting a internet connection. If the only reason why you are going to get one is that you want to see how the internet can affect your communication systems, then might as well stick to the present protocol you are using. But if you are really looking for means that you can be efficient, then SIP trunk is surely for you.

Saturday, January 22, 2011

SIP Trunking Providers and Their Benefits

SIP trunking is a unique Internet telephony service which is quite popular. It provides seamless access between the web and public telephones by putting data and voice into a single line of communication.

The SIP trunking providers are the ones who offer this unique Internet telephone service. The advantage of this process is its ability to converge data and voice communication into a single line which enables the users to access seamlessly both the public switched phone network and the Internet.

If you wish to install SIP trunking you should contact your nearest providers. The installation involves four basic components without which it will not work. To successfully implement this system you will require a SIP enabled PBX, SIP enabled enterprise edge device, Internet telephony provider and lastly, a public switched telephone network.

Therefore, you get four components such as PBX, EDGE (enterprise boarder element), PSTN and ITSP from the SIP trunking providers. The full form of PBX is private branch exchange. This is a particular telephone exchange catering to a specified office or business house. It is the opposite of a general telephone exchange which caters to various individuals, businesses and offices. The PBX used in most of the cases are IP based PBX which communicate with endpoints over the IP network. However, these PBX may well be the conventional analog or digital PBX. It is one of the fundamental requirements and an interface from where SIP connectivity is available.

Another important and necessary component provided by the SIP trunking providers is the EDGE, commonly known as enterprise boarder element. EDGE is a term which is synonymous with devices such as digital wireless phones, digital phones, softphone, etc. The local network PBX links to the Internet through the EDGE.

The SIP trunking providers are also known as the Internet telephony service providers or ITSP. An Internet data service is provided by the ITSPs for facilitating phone calls via the VoIP system. At this stage session initiated protocol initiates. The providers of Internet telephony supply connection to the PSTN or public switched telephone network for facilitating landline and mobile communication. It is the providers of Internet telephony who offer SIP trunking which enables connection between EDGE and PSTN.

Thus, SIP trunking providers are increasing by leaps and bounds especially in the US, as it is one of the easiest and most affordable options for calling purposes. It is easy to install the system at home or in the office. Some of the well-known SIP providers are AT&T, Bandwith, Checkbox, Didlive, etc.
George Yee is a consultant for sip trunking providers .

Saturday, January 15, 2011

The Best Solution

Web conferencing software has come a very long distance over the last ten years every Friday depending on the plan because a lot of progress made by the manufacturer of the device software from around the world.

At the beginning of the conference to help the world wide web on basically means that low-quality voice calls over the Internet and may also be poor media quality online video feed. At present, however, the situation completely transformed and world wide web Hot conference periods very much appear to be the real deal.

That actually ensures that a web webinar meeting between a dozen, for example, participants now offers the exact same benefits one would get if they would be basically present in the conference area. Large window screens and remarkably capable video clip imaging pretty much completely eliminates the feeling of long distance between the contributors. Due to the high speeds which might be now probable through high speed connections, the particular sound received from the additional terminals is usually as good since the quality of your respective loud loudspeakers.

The security conditions that were generally associated with any file geneva chamonix transfers and doc exchanges carried out over the internet are actually a thing of history and there is not even attempt to fear any more really.

Web conferencing software you end up purchasing must reveal all your preferences and requirements and with the great variety of products that are actually available on the market it would be virtually impossible to say you had to settle for any product that would not fit all of your needs in terms of technical functions, pricing and the like.

While world wide web conferencing software is usually designed for large meeting groups that would include close to a dozen or even more participants there’s also web meeting software applications which can be especially created for smaller groups.

It goes without saying that the latter option would be far more affordable.

This is one good reason why, before you even start studying web webinar software providers, you need to correctly establish your entire needs as well as technical specifications. It might not look like a big deal, but also in numerous international locations around the world the world wide web services aren’t in reality up to date also it would be secure to believe that the exchange speeds can be inferior towards broadband rates we have all obtained accustomed to. This basically ensures that in the case once you purchase a good quality web conference meetings software and employ it to communicate together with members which have low quality online connections, you would be fundamentally throwing significant amounts of money out your window

USB VOIP Phone Adapter, Support SIP, Skype
Cisco SPA942 4-Line IP Phone with 2-Port Switch
Siemens Gigaset Digital Cordless Phone with Hybrid IP/Landline Calling (A580IP)

Sunday, January 9, 2011

SIP Trunking business

What the heck IS a SIP trunk anyway?

OK, SIP trunks are basically just virtual phone lines, plain and simple.
They only work with an IP enabled PBX in a business environment, so aren't used in residential applications (unless you gotta LOT of kids on the phone)
So SIP is Session Initiated Protocol, this is basically Geek for a "computer language that carries your voice over an internet or MPLS connection to make a phone call". It's not that complicated once you understand what you need to use SIP VoIP:
1. You need an IP PBX, you cannot use SIP with other VoIP devices.
2. You need to have a quality Internet connection, or perhaps an MPLS WAN set up for branch to branch calling if this is your need.
3. You must have a quality SIP provider, because we get what we pay for, and we don't want to sound like Yoda or Yogi Bear.

When do we use SIP Trunking?

The main business drivers for SIP VoIP is the same as any larger scale VoIP deployment - to save money.
First off, let's not forget the fundamental way VoIP saves us money is CONVERGING our voice and internet data on one pipe, generally a dedicated T1 or larger, mission critical type circuit. Then we get to fire the phone company and they can take those old expensive lines (or PRIs) with them.
By using SIP trunks, we can many times reduce the number of "talk paths" we need coming into our facility compared to "old phone technology", thus cutting costs even further.
A by product of almost any VoIP deployment is "dynamic bandwidth allocation" this means we're delivering the VoIP over our internet pipe, and when we're not talking on the phone, the entire internet pipe is available for our use.
Additional economies of scale occur when we realize the MORE Branches we have, the MORE we can consolidate the required lines, many times bringing all the lines (SIP trunks) into a centralized location, and then firing the phone company and all their expensive lines at each location.
A good example of this is a regional bank. Ever noticed how you always call just one number, but you can get to any of the eight branches? That's because they're using SIP trunks, all the branches are connected together using either an internet VPN or an MPLS WAN and everyone is an extension off this one PBX back at the main branch.

How do we buy SIP trunks?

There are many flavors and permutations. Basically the wisest thing to do is have "one throat to choke" when it comes to this stuff.
This means, try and find a provider that will deliver a reasonably priced dedicate internet connection, and the SIP trunks all in one package. This way if anything ever doesn't work, you make one call.
Of course, you could also have an issue with your "phone guy" and the IP PBX itself, but it's much more likely to be the VoIP or internet service itself.
Other options are using a Telecom Broker. These folks really know the ins and outs (or they should) of VoIP and SIP in general, and also specific providers in your area.
Telecom Brokers shop the market, so you don't have too, and in conjunction with your PBX distributor should be able to fashion a great solution that both saves money and is every bit as reliable as your existing "old" phone service, just less expensive and more efficient!

How does SIP pricing work?

Well, again, lots of options here, and generally you get what you pay for. Most good providers will be unlimited INBOUND minutes, and then metered OUTBOUND calling. We do away with long distance charges, because every call is counted against our minutes.
Most providers will bundle say 500 outbound minutes on each SIP trunk, and then aggregate the minutes across the entire platform. So 10 SIP trunks will have 5,000 outbound minutes - call across the street or across the country.
This equates to like 83 hours OUTBOUND a month, so even if you had 20 people at the business, they would each have to be making a solid 30 minutes of outbound calls a day to use all this SIP, probably not gonna happen, unless you're a call center - and they have different pricing and needs.
Even so, if you make a lot of long distance calls, then SIP will be cheaper per "line" and also per minute anyway, shop it around, and don't forget the INBOUND is always free anyway.
Long and the short of it, don't be too worried that it's not "unlimited" outbound calling, it's virtually unlimited.
Some SIP providers call it unlimited anyway, and then tag you if you are a call center; it's in their agreement, in the fine print. Again, use a broker to determine what you need and who's the best provider for you.
Well, that's about it for SIP trunks, SIP VoIP and all that cool SIP stuff.
Go out, get yourself an IP PBX and just SIP it!
Got more SIP VoIP questions?

Saturday, January 8, 2011

SIP Trunking within your business

Why Deploy SIP Trunking within your business?

SIP otherwise known as Session Initiated Protocol has widely been used in verbal and written context with business telecommunications lingo. While this technology has been raved about and highly documented throughout the past recent years in tech blogs and websites, it can be somewhat difficult to understand exactly what SIP is and how it is used if you are somewhat new to telecommunication terminology. If you aren't a CTO or CIO but run a business with high levels of voice and data traffic, this should help better explain the technology for you.

SIP runs over an IP (Internet Protocol) network. A network running the Internet Protocol previously allowed for data transfer between satellite office locations via private lines, for hosting websites, or hosting other data servers. Over the past decade however,
new technologies
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and the explosion of VoIP (just think Vonage and Comcast phone) have given an IP
network
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far greater importance and precedence over business communications.

SIP allows for simple daily business functions as simple as phone calls (inbound/outbound) utilizing VoIP technology but can be more diverse in the business environment.


Businesses often need to protect confidential data from their competitors, such as commercial plans, corporate strategies, sales reports, organizational secrets, intellectual property and department restructure proposal. A majority of the data leakage occurs from within the organization, either on purpose or by accident.
A daily function in the enterprise or business may be more complex functions such as conference calls that additionally may require multimedia (live stream video, live document presentations, etc.) as well.

Other functions and features made available through SIP can include push to talk or chat which has gained much momentum on the web in the e-commerce world. In today's economic environment, this can easily equate into an ROI scenario due to the need to more quickly and easily allow the customer to make purchase decisions online, many times from mobile devices.
The same ROI scenario can go with the web conferencing abilities made available through this technology.

As consumers and B2B clients become more tech savvy, businesses hungry for revenues are turning to SIP to help bridge gaps between marketing mediums and the closing sales process. It is through the
convergence of multiple communications that this process can be streamlined and collaborated on.

While the above examples can be shown as uses for this technology there are many more applications that can and are being implemented in converged business telecommunications. Some of these include caller ID, IM (Instant Messaging), e-mail, and other web and voice based communications.

It is important to know that the QoS is not controlled by SIP in any way. The bandwidth and network configurations over that bandwidth as well as the carriers providing the bandwidths are just as important to the successful implementation of SIP in business communications. Other important aspects can be local hardware configurations and current in-use phone systems.

It is because of the many possibilities associated with a business embarking on purchasing an SIP solution that the use of a telecommunications broker or consultant is vital for successful deployment. Additionally, a broker or consultant can better suit your business with the proper solution or technology within your telecom budget.

This can be extremely important as some businesses are better suited for a smaller business VoIP or T-1 voice solution. A good broker will not oversell you and make sure one of the other solutions wouldn't better suit you first.


Tuesday, January 4, 2011

advantage sip trunking

Before discussing the benefits of SIP trunking, you should first know what SIP trunk is. A SIP trunk is a logical connection that uses SIP or (Session Initiation Protocol) to set up communication over the internet between a customer location and an internet telephony service provider (ITSP) which transfers the SIP calls to a PSTN (Public Switched Telephone Network). This sort of telephony connection is a boon for various businesses, especially small scale organizations that need to make regular local, STD as well as international calls and have to take a dedicated line from a telephone provider. To define it in short, 'SIP is the real-time communication procedure for VoIP phone system'.

Communication, that too effective and uninterrupted communication is one of the most essential requirements of any business today. If organizations are unable to interact and keep in constant touch with their clients, then they are going to lose out both on existing as well as prospective clients. With the latest SIP technology communication will not only be fast and efficient but also very cheap. Companies that have numerous branches scattered all over the world can also benefit immensely from this telephony system. Even if they make international calls, it would cost them the same as a local call.

With the advent of VoIP (Voice over Internet Protocol) technology, businesses can make calls over a broadband connection that allows them to make cheap or free calls, calls to various places even when they are not in their centralized location; employees can get connected from home or nay other remote area and host of other such benefits. The hosted VoIP phone system also offers tailor made and customized packages to suit every business needs. But even if the benefits are immense, many organizations deter from removing their traditional PBX systems completely, with the fear of losing control over their personal PBX system. This where the usage of SIP trunking comes in, which allows companies to preserve their traditional PBX system whilst running hosted VoIP functionality simultaneously.

The benefits of SIP Trunking are many:

• As mentioned earlier, it reduces calling cost to a great extent and you can turn all calls to local calls. Since calls travel over the internet, or through the VoIP phone system to a termination point, the charges on long distance calls are reduced.

• SIP trunking also reduces the costs on separate voice and data connections and increases the benefits for communication systems using both voice and data together.

• The capacity of this sort of phone system is huge with the potential to serve an entire organization, irrespective of its size. Big MNC's or multi-size organizations can use a single SIP trunking account rather than multiple PRI connections.

• As business grows, the communication can grow easily without having to make gateway or card investments.

• There is no more the need to use wires in bundles; communication can be transferred digitally with the help of this technology.

This is one of the smartest decisions any business organization can take. They can save money on all their communications and invest it in other areas and increase productivity. Moreover this is one of the best ways to keep in constant touch with partners, clients, employees at various locations, customers, vendors etc. What's more, this method can be used for three way calling, conferencing, traditional voice calls, instant messaging, application sharing and any other facility that a business requires to prosper.