The decision to move from “traditional” PBX to cloud telephony ( IP handsets ) or SIP telephony is not for everyone …. Or is it ?? . There are many factors to consider before deciding on the vendor , the technology , setting a budget and signing a contract and you will first need to do some company self-assessment to see how VoIP will be used.
You will have a telephone on each desk for the employees to use. Will you replace all of your phones with IP phones ? or will you use softphone applications and headsets on your employees’ computers , or perhaps smartphone APPs ?.
If you plan to use IP phones, do you have enough Ethernet ports to add phones? If not, you could consider more CAT5e/Cat 6 internal cabling although some IP phones have “pass-through” capability in the form of an Ethernet switch to allow a PC to be plugged into the phone. That eliminates the need for another Ethernet port , but only if the built-in switch provides adequate speed and throughput.
For IP desk phones, you need to know that (unlike most PBX desk phones) they require power. It can come in two forms: PoE (Power over Ethernet) from the network switch or a local PSU / power supply unit per handset. . If you choose PoE ( which we would always recommend ) you can have back-up power ( a UPS for example )in the data cabinet and the phones will continue to work even if there is a power cut.
What about your analoge devices? You may still have fax machines , alarm panels , PDQ machines etc that may still be on an analogue connection. In some instance an ATA ( analogue terminal adapter ) can be used ; which is a device that converts the IP signal to an analoge one. ; or perhaps things like fax to email .
The points for consideration are not really a significant barrier when broken down into manageable issues and your vendor should help you decision make and resolve. The benefits to your business of moving to cloud telephony , for example , ould be much more than inclusive calls and worry free operation .
Cloud based telephone systems benefit from all the “traditional” functionality of a telephone system and much more ; including call history , call recording , easy setup of multiple site location , smartphone extensions , call queues , unlimited lines , a wide range of telephone numbers , disaster contingency options , instant conferencing and call centre functionality .
An upgrade to cloud telephony is usually available for minimal CAPEX
Here at Phoenix Link we offer prospective cloud telephony clients demo units so they can “try before they buy “
So many of us seem to communicate exclusively through social media, digital apps, and live chat.
But does this really work for business communications ?
64% of SMEs said that their business could not carry on without fixed-line phones, and 52% said the same for mobile, according to research by Ofcom.
Maybe the right question wasn’t asked because what most SMEs cannot trade without is some form of internet access ( fixed line or mobile ) and with that the associated internet or mobile based voice communications….. the ability to talk.
There will always be a proportion of people who want to speak on the phone to another human being when contacting a business and many of these don’t even leave voicemails if they don’t get to talk to a person straight away.
But the ways in which those kinds of conversations happen inevitably shifts over time. Increasingly, business people communicate through conference call facilities or video conferencing services , and there is even the growing popularity of features like FaceTime ( massive in with consumers ) that inevitably is more and more used in the world of business.
SMEs will need to ensure that their telephony and internet access infrastructure can support all these forms of “ talking “.
Because small business life can be unpredictable, SMEs have become more and more reliant on flexible working scenarios – including the so called “gig economy”. ( the gig economy is defined as – a labour market characterized by the prevalence of short-term contracts or freelance work as opposed to permanent jobs.) Over the past year, two in five small enterprises in London have employed gig economy workers, as well around one in five outside of the capital. Of those who did, a quarter described as the gig economy as the “future of how businesses work.”
Whilst this allows SMEs the flexibility to better manage their workforces; it also means a larger number of remote employees will make up those workforces.
To remain efficient in this environment, voice infrastructure will have to become leaner and more flexible. SMEs will need to be able to quickly bring new short-term contractors onto their telephony infrastructure and just as easily remove them once contracts end. Similarly, if contractors are in different locations or time zones, it is likely that VoIP and video conferencing will become the norm.
SMEs must adapt to the way employees need to use voice.
Any business can be absolutely certain that Wi-Fi demands will increase but can you count on budget availability ? One way to keep growth and budgets aligned is to reduce the Total Cost of Ownership of your WiFi infrastructure. The Ruckus wireless access points that we supply , install and configure offer better capacity and coverage, often able to support 30-50% more clients than other similar WiFi solutions .
Ruckus Unleashed Access Points provide :
Simple, Intuitive Cloud Managed WiFi
Use fewer APs to cover a given area and number of users or use the same number of APs to serve more users and more traffic.
Ensure that each and every user is connected with sufficient bandwidth to support their applications.
Enable employees of all skill and experience levels to manage the WLAN.
Get WLAN management tasks done faster.
Critical event auto-notification maximizes network uptime and minimize status-checking.
Analytics & Reporting
Up to six-month network data retention enables more accurate growth planning
From application visibility to detailed RF related reports at highly granular 15-minute intervals, IT staff are able to monitor and optimize network characteristics
Manage and monitor the network from any Android or iOS device.
Shorten deployment duration by scanning APs—automatically adding them to the network.
Instantly set up guest access while away from your desk.
Dynamically insert your own advertisements (photos or text) onto your guest Wi-Fi portal at any time.
Promote your brand by customizing your captive portals.
Connect your guests quickly and easily with social login and email/SMS guest passes
Apply different network privileges to guest users on the network.
Set up dedicated and separate workflows for sponsored guest and self-help guest Wi-Fi configurations.
As a result of MiFID II regulations any organisation providing financial services to clients linked to ‘financial instruments’ will have to record and store all communications intended to lead to a transaction. With MiFID II any organisation that’s even giving advice that may lead to a trade or investment will also need to comply with this rule.
What are we doing?
The Horizon cloud telephony platform is being updated.
Horizon call recording is being expanded with a suite of integrated call recording and PCI compliance services providing a solution in accordance with all legal and regulatory compliance legislation.
If you need more information or assistance call 01227 200625 or email firstname.lastname@example.org
As a part of Gamma’s network ongoing commitment to the cloud PBX product, Horizon, they will be undertaking a migration activity of all voice and signalling traffic.
As part of the ongoing network infrastructure updates they will be routing traffic off of the network elements that are now approaching the end of their technical life. It is also a precursor to deliver some new and improved services, such as advanced Call Recording, PCI Compliancy and Horizon Meet our new Audio, Web and Video conferencing service, in 2018. These services are being developed and tested on the new infrastructure.
The migration period, which is scheduled between the 24th October and the 7th of Nov, consisting of 5 individual change control windows. Although the anticipated impact has been kept to a minimum, the process of migrating this traffic carries some risk and your service will be impacted.
During the maintenance window there are two key timeslots where service will be impacted: 00:00 – 00:15 the update being deployed on the network will cause devices to become unregistered.
Horizon Companies will be unable to receive and place calls during this time and any programmed diverts or schedules will also be inoperative. 00:15 – 01:15 devices will automatically reregister based on an internal timer that refreshes its registration once an hour.
Hence, in the worst case, a devices unregistered status may last up to an hour. Any divert or schedule setting will operate as programmed. The vast majority of Companies and users should see no real impact as all handsets will automatically reregister with an hour of the initial 15 minute period from midnight completing.
As a result of MiFID II regulations any organisation providing financial services to clients linked to ‘financial instruments’ will have to record and store all communications intended to lead to a transaction.
Also with MiFID II any organisation that’s even giving advice that may lead to a trade or investment will need to comply with this rule.
MiFID stands for the “Markets in Financial Instruments Directive”
So what are we doing?
Our Horizon and SIP Trunk offering is being upgraded with a suite of integrated call recording and PCI compliance services providing a solution in accordance with all legal and regulatory compliance legislation.
Are you fed up with having to use “walk about” wireless handsets (DECT handsets ) ?
Fed up with wireless handsets going missing when someone forgets where they have put them , batteries running out , patchy DECT cell station coverage , limited call functionality etc. ??
We have a solution .
An App for your IOS or Android smartphone ( doesn’t even have to be a work phone ; it could be one someone’s personal phone ) that is connected to your Company phone system as an extension . All your smartphone needs is to be connected to WiFi ( business, home or public ) or to mobile 3G/4G data.
The Horizon Smartphone App allows you to make and receive calls using the Horizon IP cloud telephony service from your smartphone device. This is ideal for people who often work remotely or work from different locations within a building , where it is not practical to install a desk handset or where wireless handset coverage is limited. In addition, the App also provides presence and instant messaging between users, keeping you connected no matter where you are. The App includes functionality which allows calls to be answered from lock-screen, access to Company and Personal Directories, DND , call conference , call transfer , call history and busy/no answer or always call forwarding .
Active calls are not interrupted if your device switches from Mobile Data to Wifi or Wifi to Mobile Data.
The monthly service subscription charge includes a DDI number , extension number , ability to display the office main number or DDI on outbound calls , withhold number on outbound calls , and free of charge calls to UK 01,02,03 and 07 mobile numbers.
Notification Title: Horizon Product notice for Voice Migration
As a part of the ongoing commitment to the Cloud telephony product , Horizon, the network will be undertaking a migration activity of all voice and signalling traffic. Although the anticipated impact has been kept to a minimum, the process of migrating this traffic carries some risk and your service will be impacted which is scheduled between the 24th October and the 7th of Nov, consisting of 5 individual change control windows.
During the maintenance windows there are two key time slots where service will be impacted:
00:00 – 00:15 the update being deployed on the network will cause devices to become unregistered. Horizon Companies will be unable to receive and place calls during this time and any programmed diverts or schedules will also be inoperative.
00:15 – 01:15 devices will automatically reregister based on an internal timer that refreshes its registration once an hour. Hence, in the worst case, a devices unregistered status may last up to an hour. Any divert or schedule setting will now operate as programmed.
The vast majority of Companies and users should see no real impact as all handsets will automatically reregister with an hour of the initial 15 minute period from midnight completing.
Delivers precise speech transmission. Very low processor requirements. Needs at least 128 kbps for two-way. It is one of the oldest codecs around (1972) and works best in high bandwidth, which makes it a bit obsolete for the Internet but still good for LANs. It gives a MOS of 4.2 which is quite high, but optimal conditions have to be met.
Adapts to varying compressions and bandwidth is conserved with network congestion. It captures ranges of frequency twice as large as G.711, resulting in better quality and clarity, close to or even better than with PSTN.
High compression with high-quality audio. Can use with dial-up and with low bandwidth environments, since it works with a very low bit rate. It, however, requires more processor power.
An improved version of G.721 and G.723 (different from G.723.1)
Excellent bandwidth utilization. Error tolerant. This one is an improvement over others of similar naming, but it is licensed, meaning not free. End users indirectly pay for this license when they buy hardware (phone sets or gateways) that implement it.
High compression ratio. Free and available in many hardware and software platforms. The Same encoding is used in GSM cellphones (improved versions are often used nowadays). It offers a MOS of 3.7, which is not bad.
Stands for Internet Low Bit Rate Codec. it has now been acquired by Google and is free. Robust to packet loss, it is used by many VoIP apps especially those with open source.
2.15 / 44
Minimizes bandwidth usage by using variable bit rate. It is one of the most preferred codecs used in many VoIP apps.
6 to 40
SILK has been developed by Skype and is now licensed out, being available as open-source freeware, which has made many other apps and services to use it. It is a base for the newest codec named Opus. WhatsApp is an example of an app using the Opus codec for voice calls.
When businesses understand that VoIP should be embraced and not rejected then the benefits just start rolling in . To embrace VoIP the business should really understand how it works and what they need to do to give it the best chance of working . Talking to a VoIP/cloud/hosted telephony specialist such as Phoenix Link ( www.phoenixlink.co.uk ) is a good place to start.
Voice signals , frequencies are analogue….. not digital. Voice communication can only go onto a data network if it is converted into a digital equivalent that can be converted back into “voice” when it reaches its destination.
The conversion of audible sound into data is called quantization. The process of quantizing voice is typically done by sampling the sound at eight thousand times per second, assigning one of eight bits to each sample. This yields a 64-thousand bit-per-second data stream that can be reproduced at the other end.
The additional added packet information (overhead ) required to get the voice data stream across the data network when there are several simultaneous conversations can cause issues on the network , and therefore voice is generally compressed unless there is lots of available bandwidth or the voice data network is a separate network to the general data network.
Compression as a technology in not something that is new – it has been used for many years with fax machines. Voice communication can be compressed into a much smaller data stream ( than the original ) with almost undetectable loss of quality. Compression standards are expressed as CODEC ( Coder / Decoder ) , of which a common one is called G.729 which cuts the actual voice data stream down to 1/8 of the original size (the overhead remains the same). The highest quality voice has the CODEC G722 although the most common CODEC for voice is G711 which uses 64 kbps over bandwidth ( plus overhead)
A basic element of a data network is the ability to recover from lost packets by tracking and retransmitting any portions of the data that do not arrive intact and on time. The packets that handle this are a variety called Transmission Control Protocol, or TCP. These TCP packets cannot be used for voice communication. It would result in poor quality when missing packets were re-sent and inserted into the sound at the other end in the wrong place. It’s much better to just drop any defective or missing packets and listen to the part that does arrive as expected. Remember, you have 8,000 samples per second. You can lose some and still understand what is said. To accomplish this, voice uses exclusively packets designated User Datagram Protocol, or UDP. These packets are “best effort” delivery.
A subset of UDP packets is used to further enhance the delivery of voice packets in a timely manner: Real-time Transport Protocol, or RTP. This streaming-specific protocol improves the quality of sound by giving special attention to packets that may arrive late or out of sequence (called jitter).
In summary ; as long as the data network is configured correctly , with sufficient bandwidth and the right CODECs are used there is no reason why a user would even realise that their conversations are taking place over VoIP rather than the PSTN