Future of VoIP Industry

With the internet becoming one of the most used facets of business today, VoIP has gained a huge foothold in the world of Internet. While there are always improvements to be made in all systems, and we see this with the evolution of Windows products, and Apples iTunes, voice is not to be left behind.

While the protocol for voice will not change much, it has become more secure, and with each passing day there are more and more applications written to enable extra functionality out of your VoIP systems.

VoIP Telephone

Dig Deep with CDR:- From CDR (call detail recording) which can be as simple as utilizing the existing database the server uses, to creating and using an entirely different database, and importing data. Adding graphics, and a sleek interface to the data, this can be a simple tool to deploy with powerful data that can show the times of your highest call volume all the way down to what countries called you, or who is calling which country and when.

Extension Managers:- Extension managers give you more flexibility and ease in managing user extensions and can even allow you to associate specific users with specific extensions, and allow them levels of access to all or specific extensions.

IVR Trees :- IVR Trees have been in use since before VoIP, but with VoIP, we are able to do so much more with IVR Trees, as well as music on hold, or even a simple message or advertisement for callers sitting on hold, or in a queue waiting on their call to be answered. Now, while we can do all these things to an IVR, the power is not actually behind your VoIP system, but the server it runs on, and as we progress the market for applications to interface with VoIP systems increases, and your choices grow.

Just as when Apple decidedly changed the music industry with the release of the ipod, and expanded with the iphone, the movement within the VoIP industry, while not as high profile as Apple or Windows events, is always changing and evolving, and the room to create your own custom system from what we like to describe as modules is almost never ending

As more and more companies choose to utilize VoIP, the demand for utilizing aspects of the system like Voicemail to email, and faxes to PDF, will increase the demand for more robust software and reporting tools for business to track the ways in which the system is utilized, and I can foresee the future where portions may even be monetized into salable ad space to help generate revenue for the companies that use the system.

While SIP trunk providers are limited due to the finite numbers available, service providers and the offerings will continue to grow, offering companies who do not want to install, run and manage their own voice servers, all the services wrapped into a bundle right along with the DID’s for the company. The vast amount of data that can be gathered via VoIP, will be of great value to all companies in making marketing or financial decisions, and due to that the growth in how that data is gathered and displayed is going to provide a huge growth in the IT field as a whole, from programmers and developers to hosted solution providers and techs at all levels.

Limitations of VoIP!

limitations of VoIPVoIP is much discussed technology around. We hear everyone talk about benefits and opportunities with VoIP. But one thing which is not much discussed is :-

 

Does VoIP have any limitations? If yes then What kind of limitations?

 

There really are few limitations, VOIP is one of the most flexible phone systems out there. Your only going to limit yourself based on the money you want to spend, and how deep you want to dive into configuring your own system.

Probably the biggest obstacle you will run into is your internet connection, and if you utilize it for data and your SIP trunk, you can run into latency issues, however this is not something directly related to VOIP. This would be more of a 3rd party type issue, and if you are aware of what you will need when it comes to the connectivity side, you should not have any issues.

On the VOIP side, there are so many flavors and distributions as well as providers, the possibilities are almost endless. While choosing a hosted solution, you will have limits on how many conference rooms you might be able to have, you may not have the option to forward voicemail to email, your IVR tree may be limited, but again, for more money, you can always remove those limitations.

When configuring your own VOIP system, and simply outsourcing the internet connection and SIP trunk, you can remove just about every limitation. Again, not all of them, but the ones that are left again, relate to hardware and money, as well as office footprint. Current systems can handle 100+ calls simultaneously, are you willing to spend the money on the switches and handsets to handle 100+ handsets? Can you have your ISP provide the needed bandwidth? Is there room in the office for all the equipment?

Typically we see between 40-50 handsets in a typical mid-size business. I currently have a customer with just shy of 50 handsets, spread throughout the United States, and 1 single server located in Chicago. The system is extremely flexible and proven solid and reliable. Not only will your own system handle the calls, IVR trees, and ring groups, voicemails, it can, and when configured, alert you to updates, provide reports on calls, times of calls, inbound/outbound calls, to what countries etc. You don’t have to wait on a invoice from a provider or request the information, all of that is contained within the system itself.

The other limitations you may actually have are more than likely software dependent. While I utilize open source PBX systems, running over a Linux operating system, there are proprietary systems out there, such as Cisco, and Microsoft. The fact is you will more than likely be utilizing support when using those types of systems, and not running or installing on your own, but, again, you will hit limitations imposed by the company your dealing with. And it is not that these are deal breaking limitations, but having spent the last year and a half administering phone systems, the limitations are in place to ensure quality service, and reliability, not to hinder use of the system. I should also point out, that when paying for this service, while you may have extra charges to increase conference rooms, or increase and IVR tree, or alert to voicemails via email, most will supply a far more user friendly detail records available in a web interface.

Choosing to use a provider, or to do it yourself, understand and realize, that this is not, as with anything going to have 100% up time. While your side, or the provider can remain up, there are so many other places for issues to arise as well. Switch’s, core routers in major data centers, the receiver of the calls you make, while these remain very scant in occurrence, they can happen, and it does not, and should not reflect negatively on VOIP systems. Today we are heavily reliant on the internet and network connectivity, and will always face challenges to keep the information flowing, but in the world of phone systems, VOIP and SIP trunks are flexible and reliable for all of today’s business!

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What is Codec?

A CODEC is an abbreviation for coder-decoder. Its a device that encodes or decodes a digital stream to transfer it over data network. Two basic operations of CODEC are:-

To convert an analog voice signal to its equivalent digital form for easy transmition.

Once the transmition is complete it converts the compressed digital signal back to its original analog form.

There are quiet a few audio CODEC’s available in the market. Some you will need to pay for and others are available for free. Each Codec uses certain network bandwidth and based on the computation determines the sound quality.

Table below lists the most commonly used Codecs in the VoIP telephony.

Legend for table:

CODECIndustry name of the Codec
Bandwidthlists the bitrate in kilobytes per second for each Codec.
MOSMean Opinion Score measures the perceived quality of audio after it has been compressed, transmitted, and decompressed by the particular codec.
Measured on the scale of 1 – 5. 5 being best
Description: Gives basic explanation about the Codec.

CODEC Bandwidth
(kbps)
MOS Description
G.711 64  4.2 Free Codec.
Consumes maximum bandwidth as compared to other Codec’s
G.711 uses 64 kbps for one way transmission. To complete a two way call it uses 128 kbps. G. 711 has two variations A-law and u-law.

A-Law: used in Europe
u-Law: used in USA and Japan

This is codec is also used by PSTN and is known to provide the best voice quality.
It has poor network efficiency
G.711.1 is an extension version of G.711, It allows the addition of narrowband and/or wideband

G.722 48/56/64 4.1 Free Codec
Wideband Speech Codec
Significant improvement in speech quality as compared to G.711
G.723.1 5.3/6.3 3.7 – 3.9 Free Codec
Has two variants.
First variant has bitrate of 5.3 kbps MOS 3.9
Secont variant has bitrate of 6.3 kbps MOS 3.7
Very high compression while maintaining high quality audio
Lower quality than many other codecs at similar data rates
G.726 16/24/32/40 3.5 – 4.1 Free Codec
Transmits voice at variable rates of 16, 24, 32, and 40 kbps
G.726 was designed to supersede G.721 and G.723
Frequently used for international trunks in the phone network
Not well-suited to music or sound effects
 G.729  8  3.92 Paid Codec. Require specific VoIP phones/gateways to implement this codec.
Requires low bandwidth
Good Voice quality

 

Why Single Tier SIP Trunking is the best option

Most of the voip service providers offers multitier SIP Trunking plans. This is not only confusing but also turns out to be atleast 25-30% more expensive to most consumers. Lets review by an example. Say a company needs local telephone numbers all over US to provide local reach to their customer. Toll free is obviously an option, but the cost of owning a toll free number is 10 times more than local access numbers. Not only that Local access numbers also called DID (direct inward dialing) numbers improve your company’s visibilities in local areas, but also work better for search engines as compare to toll free numbers.

So for most companies traffic is never the same for 24 hours. There is always more traffic during some hours as compare to others. Here is sample usage for a normal calling card business. Most VoIP service providers offer Tiered rates. Tiers 1 for East Coast states, Tier 2 for central USA and Tier 3 for West zone. So this business has peak traffic in morning around 9:00 and 8:00 PM in each zone. So morning 9:00 AM in Newyork is 6:00 AM in CA. By the time traffic increase in CA, you see a decline in traffic in Tier 1, East Coast. But the service provider is going to charge you for the peaks in each zone. Notice 500 Channels (tier 3) usage at 12:00 PM (which is 9:00 AM in CA). At the same time Tier 1 usage is only 300 Channels. You are not allowed to use the unused capacity in one tier to the other tier. You have to pay for the total of 1500 Channels (@ 500 Channels peak in each zone). While at DIDForSale we calculate the usage based on total of all the zones at any give time. We dont charge based on max of each zone. With our largest single tier coverage in US, our customers pay almost 30% less.

Time EST Tier1 Tier2 Tier3 DIDForSale Total Tier Commitment
1:00 AM 100 100 100 300 1500
2:00 AM 100 100 100 300 1500
3:00 AM 100 100 100 300 1500
4:00 AM 100 100 100 300 1500
5:00 AM 100 100 100 300 1500
6:00 AM 150 100 100 350 1500
7:00 AM 250 150 100 500 1500
8:00 AM 350 250 150 750 1500
9:00 AM 500 350 150 1000 1500
10:00 AM 400 500 250 1150 1500
11:00 AM 300 400 350 1050 1500
12:00 PM 300 300 500 1100 1500
1:00 PM 300 300 400 1000 1500
2:00 PM 250 300 300 850 1500
3:00 PM 300 250 300 850 1500
4:00 PM 300 300 300 900 1500
5:00 PM 300 300 250 850 1500
6:00 PM 350 300 300 950 1500
7:00 PM 500 350 300 1150 1500
8:00 PM 400 500 300 1200 1500
9:00 PM 300 400 350 1050 1500
10:00 PM 300 300 500 1100 1500
11:00 PM 150 300 400 850 1500
12:00 AM 100 150 300 550 1500

This is how the Graph looks like. Short duration peaks in each zone cost you more and wider peak as compare to DIDForSale

Tiered SIP Trunking Comparision

Feel free to contact us to prepare a FREE sip-trunking report for you to see how you can save in your VoIP Bills. DIDForSale offers largest single tier coverage  in US, UK and Canada.

 

 

Asterisk DTMF issues

Its a common issue with asterisk as it sometimes wont pass dtmf properly. To solve the issue first we need to check the network have sufficient bandwidth, if bandwidth is sufficient then we need to add below parameters on all didforsale trunks. Note that if you are using G729 codec then inband wont work. First try with dtmfmode=auto, if it didn’t work then try with dtmfmode=inband.(For only ULAW) codec. If you are using g729, then force rfc2833 instead of inband.

Example

[didforsale1] type=peer
host=209.216.2.211
nat=yes
canreinvite=no
disallow=all
allow=ulaw&g729
context=from-trunk
dtmfmode=auto or (use inband for ULAW / Use rfc2833 for G729)

 

Hope this will solve the issue