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
Demand for faster, more reliable internet connectivity has never been higher for business users but the reality for many rural businesses is that fibre broadband is not yet available and Ethernet leased circuits are too expensive and with long lead times. Companies are turning to 4G for both interim and longer term solutions, and here at Phoenix Link we provide 4G solutions on a rolling month contract basis
SIMs are monitored and cost-effective data bolt-ons added as required.
For some applications, such as an SMTP, IP CCTV, ATM, M2M and PoS, that require fixed IPs, fixed IP SIM cards with public IP address assignments are available.
Small to medium sized companies are buying air-time and handsets separately and the SIM-only market has rocketed. The price of mobile data has dropped more than 70% in the last year. Some sim only resellers like Phoenix Link have access to sophisticated online tools that allow us to activate stock SIMs, track usage, set usage thresholds, SIM-swap and add bolt-ons without having to go to the mobile network to set up the order . So if you want the personalised customer service experience that you get with fixed line and cloud communications for your mobile needs ; then come to Phoenix Link for your Vodafone and O2 requirements. Email email@example.com or call 01227 200625
The Markets in Financial Instruments Directive, commonly known as MiFID II, is a set of sweeping reforms for the financial services industry which mandates the ability to record conversations relating to financial selling. As a result, network based call recording and storage will be central to the new MiFID II provisions ; timetabled for introduction on January 3rd 2018.
The Financial Conduct Authority (FCA) currently mandates that only the telephone conversations of individuals directly involved in trading need to be recorded, but the new legislation, MiFID II, will broaden the scope considerably.
The SIP and cloud telephone services provided by Phoenix Link will be compliant. For more information EMAIL firstname.lastname@example.org
It is rather obvious that combining voice and data on an existing data connection ( already connected to every desk ) is a no-brainer that saves money as long as the connection is sufficient. Having the voice traffic simply added to your existing data stream on your existing structured cabling completely eliminates an entire wiring infrastructure and eliminates IT involvement in moving a phone….. just make sure the internal switch the phone is suitable.
How can this be achieved ?
Every tiny piece of data that travels across your network in a “container” called a packet. Packets are tiny amounts of information that take many forms and carry all kinds of information. For example :
Your email arrives in packets.
When you click on a link, packets are sent.
Every picture that appears in your browser arrives in packets.
These packets can travel a long way from various sources around the world to reach your computer. As a matter of fact, two packets that are parts of the same sentence from the same origin can take utterly different routes, through different countries to reach you. When the second one gets there before the first one, it has to sit and wait for the first one to arrive. That delay of the first packet is called latency. Latency is delay in the network. When you have multiple people on the network moving large files at the same time, delays can be considerable; they can even take a whole second to arrive.
When latency occurs, a part of your network may have to increase a buffer that effectively establishes on purpose a delay in presenting subsequent packets. That purposeful delay to give more time for packets to arrive is called jitter. Whilst this works well and invisibly for such communication as email ,when jitter exceeds 150 milliseconds for voice traffic, people notice the delay and are frustrated with an unworkable disruption of the normal rhythm of a voice conversation.
Sometimes packets do not arrive at all. That is called packet loss. Some programs / protocols will request a replacement, creating further delay.
To compare these terms to everyday business life, think of department meetings. The meeting can’t start until everyone arrives. Delay is when one or more of the members is late. Over time, the 10:00 meeting seems to start at 10:15 because someone is always late. That’s jitter. If some people don’t arrive at all, or if you have to send someone for them, that’s packet loss.
Such delays are devastating to voice traffic on your network, just as they are to your meeting.
Your network may have locations with a lower bandwidth, such as a separate building or remote location. A connection that was designed for carrying email and other business traffic is likely to be completely inadequate once voice traffic is added to it. As a result, voice conversation is impossible. Hence our emphasis here at Phoenix Link as the suitability and sufficiency of the data connection.
Further, maybe that remote location has a “shared” service where your traffic is mixed with other companies’ data, or even worse, over the “public” internet. There may be no guarantee that packets will arrive at all, much less arrive on time and intact. Here at Phoenix Link we only provide business grade data connections
So moving to a VoIP based voice solution can be a major step forward or a significant pain depending on who undertakes the implementation ; so choose Phoenix Link for a smooth , painless transition
A recent research study revealed that 86% of organizations cite employee efficiency and productivity as the leading internal use of telecom technology with improving communications between employees and customers coming in second at 64%. Both use cases are indicative of alignment between technology and business objectives. Employee efficiency and productivity affect corporate expense, and communication between employees and customers influences customer satisfaction and customer retention leading to positive revenue returns.
The writing is on the wall for ISDN… how will the ISDN “switch off” by BT/Openreach in 2025 impact your business ?
ISDN as a technology ,and therefore as a telephone line for your business telephone system, is dying ( due to be switched off in the UK in 2025 ) and companies are turning to SIP trunk telephone lines as their preferred solution going forward . A SIP trunk is in essence a form of VoIP ( Voice over IP ) telephone line that connects your on premise telephone system ; via the internet (using a form of secure VPN ) rather than via a BT telephone ISDN or analogue line .
Perhaps you think that SIP trunking is too expensive and too much hassle to implement. Fortunately these concerns are unfounded in most cases and the reduction in line rental charges and call charges and the increase in usable features far outweighs any short term change management issues.
If your telephone system is five or more years old then bypass SIP and go straight to Cloud Telephony as this has the potential of reducing your costs even further , future proofing your business telephony and adding a raft of productivity enhancing functionality.
2020: Five years before ISDN lines will be switched off; businesses will no longer be able to buy any telephone systems that use these networks. Although 2025 may seem a long way off, 2020 is less than five years away…
If your business is ready for a well managed telephony change then contact Phoenix Link at email@example.com or 01227 200625 for advice on how to proceed.