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)