Posts Tagged ‘Small Business Voice’
Many small businesses are moving their phone systems to VOIP PBX. VOIP PBX Systems provide significant savings on monthly basis and offers enterprise level features to small businesses. There are many great companies which offer competitive VOIP PBX systems including Fixtro. Vendors offering state of the art, next generation voice solution, talk highly about the value of VOIP PBX, display the incredible features and impress customers with significant savings over their current phone bill. During this magnificent sales presentation, vendors ask, “How many phone lines do you need? And how many minute they will use?” However, most vendors never ask the most important question to their customers, “Is your network VOIP ready?” This critical question is often overlooked because vendors are afraid to lose the sale. They hope that client network is VOIP ready and customer is locked into long term contract before any problem is discovered.
This strategy can result in huge problem for companies who rely heavily on phone calls for their revenue. You are guaranteed to have phone problems if your network is not VOIP ready. Therefore, it is very important for you to know in advance if your network is VOIP ready and what can you expect once your phones are installed. You want to know the problems before they occur and resolve them before implementing your VOIP PBX.
You need to ask these 5 Questions to help you determine if your network is VOIP Ready:
1. Are the WAN and LAN circuits capable of handling the increased traffic from the VOIP deployment?
- You want to know your network throughput.
- You want to know what applications are using what percentage of bandwidth.
- You want to know how much bandwidth you have.
- You need to know if there are any errors on your network port in router or switches.
- Is there any kind of excessive utilization on any ports?
2. Do you need to implement QOS (Quality of Service) polices?
I recommend QOS in almost all networks but if you have enough bandwidth you may not need it. However, if you are on border line of bandwidth requirement than you will need to implement QOS. Older switches and routers may not support QOS. Therefore you will have to upgrade your equipment. Therefore, Question #2 becomes even more important. You must understand how your network bandwidth is currently being used and if it can handle the extra VOIP traffic.
3. What type of mean opinion score (MOS) call quality can you expect on your network?
Explaining a call quality to someone can be somewhat tricky. Therefore, experts came up with a numeric value that is given to call quality of phone call. This numeric value is known as mean opinion score or MOS.
MOS score varies for each phone call. Therefore, it is important to measure it as an average so that you can set realistic expectation and come up with a base line for your VOIP network.
4. What factors can affect the voice call quality – delay, packet loss, jitter, codec?
- Packet Loss – Packet loss happens when packet is dropped by switch or router due to some sort of error. In most applications, dropped packets are retransmitted but due to real time requirement of VOIP calls, voice packets are not retransmitted. This results in gaps in speech, which will drop call quality.
- Delay – Packets are sent based on first in first out. Due to this rule, packets are stored at router or switch for their turn to be transmitted. A voice packet not prioritized to be sent first when it arrives at the switch or router will suffer a delay and degrade the voice quality.
- Jitter – Jitter is defined as variation in inter-packet delay. Voice packet is sent with 20 mille second gap in-between packets. If you network has delays at various points than 20 mille second gap can easily turn into 100 mille second gap. Some jitter is expected but excessive jitter can be a problem for your VOIP network.
- Codec – Various codec’s sample the voice at multiple intervals. You need higher number of voice sampling in order achieve better sound quality. Most common codec used in industry is G.711 and G.729. G.711 codec takes more samples than G.729 and therefore results in better quality phone call.
Packet loss, delay and jitter may change during the day, week or month. Due to this change, call may be OK on Monday but may not be ok on Tuesday or call may be ok in morning but bad in afternoon.
In order to avoid these factors, complete testing of network should be done during various times to get good understanding of bigger picture.
5. How many concurrent VOIP calls can your network handle?
Well, it depends upon the available bandwidth and codec used for the voice calls. I guess better question to ask would be: Do you have enough bandwidth?
To come up with required bandwidth, you just need to guess how many simultaneous calls will be placed across the WAN link and multiple that numbers by the bandwidth metric per codec. For example, G.711 requires 64Kbps per call. Meaning, you can get (64 kbps x 24 = 1.5 mbps) 24 calls on a full T1 Connection. Another Codec, G.729, only takes 8 Kbps per call. G.729 codec allows four times more concurrent calls on same T1 connection.
Let’s say your office has 20 users and 5 people will make simultaneous phone calls to remote office. Your required bandwidth for VOIP network will be 5 x 64 Kbps (G.711 Codec Used) = 320 Kbps. This means you need to make sure you network has at least 320 kbps of extra bandwidth available during normal business hours.
Gathering detailed answer to all these questions can take significant time depending upon the size of your network. But this research will save you time, money and headache over the long haul. Your superiors will appreciate your good judgment and attention to detail.
However, answering these questions is beyond the scope of normal computer user. In fact, they are beyond the scope of average VOIP Company. I have personally called at least 10 VOIP PBX vendors and not a single company asked me, “If my network was VOIP ready?” Most of these companies, quite frankly, don’t care if my network is VOIP ready. And the ones, who care, didn’t bother to ask. This tells me, VOIP Service Providers are expecting that their customers ensure the VOIP readiness of their network. In most cases, do not expect any help from your VOIP Service Provider if the problem exists with in your network.
I have a feeling, you might be wondering, “Can Fixtro help us determine, “If our network is VOIP ready?”. Answer is yes, we can. We can help you gather all the answers and help you carefully design a solid VOIP network.
Hosted PBX systems are disliked by some small business owners. They opt to have some hardware in their own office under their control for the money paid. Further small business owners who want less than 20 connections do not have anything to do with multi-site support and other advanced features. All they want is a PBX system with limited features and low cost, suitable for their business. Cisco Small Business Unified Communication 320 is a recent PBX system released by Cisco. The system is appropriate for small offices and small businesses.
The hardware looks like a small router we use for internet connection. The box has four Gigabit Ethernet ports, 12 FXO ports, Virtual Local Area Network (VLAN) support and SIP trunk support. There is integrated voice mail and auto attendant software coming with the package. SIP trunk manages to get rid of all the heavy instruments and make the hardware required for PBX systems as small as possible. Hence, Unified communication 320 is known for its small size and high productivity. Unified communication 320 is by far the cheapest PBX system in the market. What makes it more irresistible is its brand name.
All you have to do after purchasing the hardware is to connect it to a 24 port switch and start using the phone lines. All the data, voice and multimedia needs of the company are taken care of by the PBX system ensuring customer’s full satisfaction. Hardware, switch and software, what do we talk to? Of course the companies should connect separate telephones to the switches. But the system supports low cost Cisco SPA300 and SPA500 SIP phones. This makes the system much more affordable. Small offices looking for PBX systems which they could control themselves find Unified Communication 320 as the best choice. The system received a lot of positive reviews stating its performance is much better than its other counterparts from Cisco.
Fixtro provides full assistance in assembling the Cisco Unified Communication 320 PBX system. We also provide the necessary customer service required in case of any repair. Customers who purchase a full Fixtro Unified Communications system get the Cisco Unified Communication 320 PBX systems for free. Huge companies who get more than ten lines can make use of this free PBX system to manage a smaller branch or a division of the office. The offer is valid for a limited period of time only in Fixtro.com. Only customers of Fixtro.com get a chance to try their hands on the Unified communication 320 PBX systems for free.
Want to learn more about Cisco UC 320, Give Fixtro a call at 510-709-4060 and schedule a free demo. We will show you the full power of Cisco Unified Communication Device (AKA UC320). You will able to experience the truly state of the art phone PBX system. You will learn about Cisco Phones, UC320, Smart power of Ethernet Switches that will make your company look like next fortune 500 company.
Don’t worry. Consultation is free. Pick up the phone and give us a call today 510-709-4060.