If you have begun planning VoIP implementation for your business, then you have come to the right place. You will need to evaluate the various aspects of your core network, uplinks, VoIP connectivity to legacy PSTN systems, ISDN circuits and calling record.
Network for VoIP implementation
Various businesses deploy VoIP systems without proper planning which can become a major problem when your business expands and you start to experience the capacity issues. Therefore, it is very important proper network capacity planning is considered before implementing a VoIP infrastructure for your business.
In order to cover every aspect of VoIP capacity planning you will need to consider following.
- How many employees in each office
- Number of simultaneous inter-office calls.
- Average total number of inter-office calls everyday
- Average total number of PSTN / Mobile calls everyday
- Total number of International PSTN / Mobile calls everyday
- Your vision about business expansion over the period of 10 years
- How your business will expand every year and where do you see your business in 10 years time.
- Do you want to keep VoIP separate from data network or you want to implement VoIP among data traffic.
- What features do you need for VoIP?
- Are you considering VoIP recording
- Budget for the VoIP network infrastructure implementation
- Do you employees need Extension Mobility?
Capacity and size of the VoIP Infrastructure
For example if you have 4 office branches dispersed over different continents, you should consider individual VoIP infrastructure for each office for resilient VoIP system.
Have a central VoIP system can cause serious problems when there is network / link outages between offices therefore it is important that you consider having individual VoIP system for each office.
For example, if you have average 100 employees in each office and you don’t want to spend as little as possible on infrastructure then consider deploying Asterisk IAX / SIP based VoIP solution which is a Open Source solution for VoIP. Otherwise if you have sufficient budget then consider deploying Cisco Unified Communication system.
Asterisk has been developed and designed by Mark Spencer of Digium which is offered under GPL / GNU license and there are no associated costs. Asterisk is capable of delivering enterprise / carrier grade Voice along with unlimited number of features.
It is wise to separate your VoIP traffic from rest of the data traffic. Therefore, based on the number of employees in each office consider how many network switches you will require, 100 employees will probably require 3 x 48 port network switches, which will give you sufficient spare network ports for expansion. You can always bridge more switches when your base of number of employees expands.
VoIP Core System Design
Since we have considered using Asterisk as VoIP core, the best option would be to use a cluster Asterisk system to provide resiliency. Choose best of the breed servers such as HP Proliant G5 or G6 servers with dual processor.
It is very important that you choose powerful server machines because loads of CPU and memory power will be required to encode, decode and process the voice traffic. Resilient Asterisk will help in case if one Asterisk server fails other server will keep your office VoIP functional while you are fixing the failed server.
Also consider using Gigabit multi-bonded NIC (Network Interface Cards) to provide high throughput to process high number of simultaneous calls.
VoIP Handset deployment
These days there are many brands out in the market capable of supporting both IAX / SIP / MGCP and SCCP protocols. According to the available budget choose the best suited SIP or IAX phone for your employees.
VoIP delivery method planning
If you have decided to keep both VoIP and Data traffic separate then it would be wise to utilize the high payload codec’s to deliver carrier grade voice. It is highly recommended that you use either G.711 aLaw / uLaw VoIP codec’s. G.711 codec’s utilizes industry standard 64 Kbps of data bandwidth while maintaining Quality of Service.
It is also highly recommended that you use UDP on network layer to deliver your voice by using G.711 codec’s to ensure delay less conversation. Some times by utilizing TCP on network layer cause redelivery of lost packets which can cause noticeable delay in VoIP conversation.
VoIP Inter-office trunk planning
Based on the number of inter-office calls and the available inter-office bandwidth, choose the most appropriate VoIP codec’s. For example, you are utilizing G.711 codec’s for the inter-office calls and your office makes 20 simultaneous inter-office calls then you looking at 64 x 20 = 1280Kbps (1.2 Mbps) of dedicated network bandwidth between each office.
As we all know that the international bandwidth is very expansive, it would be wise to utilize the VoIP codec’s which utilizes low payload while maintaining the quality of service (QoS). G.729a/b/c codec is capable of maintaining constant stream of data between peers and utilizes only 8 Kbps payload and delivers GSM grade voice quality. 20 simultaneous inter-office calls by utilizing G.729 codec will result in 8 x 20 = 160 Kbps bandwidth which will not cost your business much.
VoIP Security Intrusion Prevention
Your voice which is a analogue signal is converted into digital signal and carrier over in the shape of data chunks which can be decoded back into voice by anyone who has got access to that data. Therefore it is very important that you consider securing your voice traffic.
The best option to secure your VoIP traffic would be to secure it on network level by encrypting data traffic. Data traffic can be encrypted by using strong encryption algorithms, consider deploying network firewall as a gateway in each office.
When VoIP traffic leaves your office firewall will encrypt it so no one can decode what is being carried over within encrypted data tunnel, as soon as the encrypted data arrives at the other end it is decrypted into VoIP traffic. I hope you have grasped the security bit therefore we will not go into further details of network security.
VoIP Connectivity to PSTN and Mobile networks
Usually VoIP systems are connected to PSTN (Public Switched Telephone Networks) via using ISDN circuits, you will need to plan how many simultaneous calls each office will be making to PSTN network. For example, if you think there will 10 simultaneous PSTN inbound and outbound calls then consider using ISDN 30 circuit. It is also known is PRI (Primary Rate Interface), E1 European standard PRI and ISDN30, it is capable of carrying 30 simultaneous inbound / outbound calls.
DID (Direct Inward Dialling) VoIP Planning
You will need to make a decision how you want to manage all inbound calls in each office. Do you want to divert all inbound calls to a operator who then diverts the call manually to desired internal extension or do you want the calls to reach each person directly.
The best option would be to deliver the call directly to each person by utilizing DID / DDI. This is a standard Asterisk feature. Based on the number of employees and accommodating near future business growth needs decides how many DID numbers you will require. For example you have 100 employees in each office consider getting 150 or 200 DID / DDI numbers from your PRI / E1 or ISDN 30 circuit provider.
For example if you are based in London - UK, 200 DID numbers will look like following.
0044 (0) 207 424 2200 - 2400
0044 is the country code, (0)207 is the city code, 424 is the exchange prefix and first DDI number is starting from 2200 and the last DDI number is 2400.
Internal VoIP extension planning
Once you have got the DDI / DID number allocation from your ISDN30 circuit provider, you will need to configure your internal extensions according to the assigned DID / DDI range. As mentioned in the above example the DDI range is 2200 - 2400, your first internal extension will be 2200 and the last extension is 2400.
For example, an employee John Matthew's internal extension is 2219, if you want to call John Matthew from your mobile, you will dial following number 00442074242219. Call will be received by Asterisk server over the ISDN30 circuit, Asterisk will read the last 4 digits which is 2219 Asterisk knows this extension and it will connect the incoming call to internal extension 2219 and John's phone will ring.
If you are not planning for DDI you can configure your Asterisk server to divert all incoming calls from ISDN30 circuit to an operator.
VoIP Call Recording
It is now possible to record internal, external inbound / outbound calls by using Asterisk. You can also configure Asterisk to explicitly record calls from certain extensions. You can also listen to recorded calls via a web browser.
VoIP Call Detail Record (CDR)
Asterisk stores data for all incoming and outgoing calls in the database which you can use to monitor the performance of your VoIP system or for capacity planning.
VoIP least cost call routing
If you have deployed VoIP in multiple offices across the globe then it would be wise to implement least cost routing for the international PSTN calls. For example, if you call a client in Japan from USA it will be considered international call hence the heavy bill. Assuming that you have another office branch in Japan which also utilizes VoIP system, you can configure your asterisk server to deliver all Japan PSTN destined calls via asterisk server based in Japan, this way call will be considered as local call from your Japan office to Japan PSTN.
Live VoIP Monitoring
There is plenty of monitoring plug-in freely available for Asterisk which you can use to monitoring live status of all internal, external, inbound and outbound calls of all your office.
VoIP Fault Tolerance and Alerting
I assume you are not a computer expert, you can get some one to write few scripts for your VoIP systems to alert you via email or a phone call when something breaks or when the volume of calls reaches certainly threshold.
Fault altering and logging is highly customizable according to your needs, just use your imagination what you want to monitor and setup up alerts.
Conclusion
VoIP is a must to have service for every individuals and business, you can build your own telecommunication system for fraction of a cost and the savings you make by utilizing VoIP are incredible. Therefore it is highly recommended that you visit our Business VoIP providers section to see what offers are available to you and how much you can save.
Fill in the below Free no obligation Business VoIP Quote form and we will let you know how much you can save.
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