Articles in the ‘Voice’ Category
This tutorial is for you if you are using DSL connection with Cisco ASA5505 and using hosted VOIP system. You are also experiencing bad call quality. More than likely its the person on the other end of the call who is having hard time hearing you. This is because your router is sending out packets using First In – First Out. If we can tweak the router so that VOIP packets go out first than you will have a much better call quality. I am going to show you how you can do this using Cisco ASA 5505.
First of all, lets do a bandwidth test to see what kind of bandwidth we are getting. I used the bandwidthtest.net
Common mistake I find in VOIP Networks is that QoS is being done using SIP protocols. SIP is only used as call control. RTP is being used while call is in session. Therefore, I recommend using RTP to prioritize your Voice Traffic. I used the port numbers 16384 to 32767 to prioritize my RTP traffic. These port numbers are used by Cisco for RTP traffic. So, basically I created a new class VOIP and matched the RTP traffic. (As Shown on Left)
Than, I created Policy Map VOIP to gave it a priority. This Policy Map will tell the ASA What to do with the class VOIP. In other words – now RTP traffic is identified and ASA knows that it needs to give it a priority.
Now, I create another Policy Map named QOS. I will use this policy map to shape all other traffic. You can do using class “class-default”. This class is already prebuilt. I simply shaped it to 280,000.
Why 280,000 ? This is not a magic number. I did a bandwidth test on DSL network and found that my upload speed is about .63Mbps or 630Kbps. I wanted to ensure I can make up to 4 calls without running into VOIP trouble. Each call takes about 80K. 4 calls will take 80 x 4 = 320K. So I need to guarantee 320Kbps of bandwidth for my calls and save the rest for all applications. 630 – 320 = 310 kbps. I reduced it to little less (280K). This is because you can only apply the average shape in the multiple of 8000. This is the limitation fo Cisco ASA 5505.
In my Policy QOS, I also created an exception for my VOIP traffic by adding service-policy VOIP comman. Basically, this tells the ASA to limit all traffic to 280K except the VOIP traffic (which is RTP traffic).
Last step left is to apply the policy to interface. We do this using service-policy QOS interface outside.
Result: Please note – there is no drop in VOIP traffic
Now, lets a do a bandwidth test again to see our internet bandwidth is limited to .28 Mbps or not. I will use the bandwidthtest.net again.
After applying the policy, I uploaded a video on youtube to see how this impacts my voice quality and here is the result:
How can you test to see if QoS is working ?
You can use the following commands to see if QoS is working.
- Show service-policy
- show service-policy shape
Here is a quick basic Video Tutorial of Cisco SPA 504G Phones. I have covered some of the basic features of this phone and our service like
- Putting the call on Mute
- Putting the call on hold
- Forwarding all call to different number or extension
- Putting the phone on “Do Not Distrub” Mode
- Tranfering Calls to different extension or number
- Going to VoiceMail
- Going into the Menu of phone
- Finding IP Address of the phone
- Resetting the phone
This video covers the basic features of Cisco SPA 504G Phone
How to Forward All Calls to number or different extension.
You can use this feature when you are leaving the office or simply do not want to take the calls.
Fixtro is your ideal service provider for installing business-class networking and voice communications across your entire company. We understand that your phones are very important for your business and therefore we recommend Cisco Unified Communication devices for your voice needs.
Affordable and easy to use, the Cisco UC320W Unified Communications lets you take advantage of IP telephony to reduce costs, boost productivity, and create a more collaborative company. This one device combines both analog and digital pbx system in one box. You can take advantage of old analog telephones as back up and reduce monthly cost by using digital sip trunk.
You can connect Cisco UC320 with up to 4 analog lines and multiple sip trunk providers. This feature ensures that you have continuous connectivity. We can configure Cisco UC 320 so that all your outgoing calls are routed via sip trunk and incoming calls come from analog lines. This format ensures your phone service is up and running at all times.
Cisco UC320 also includes wireless access point, plus advanced phone features such as voicemail and automated attendant. This solution connected with Cisco Adaptive security Appliance can make a robust, secured and redundant telephony network.
Now you can replace your PBX or key system with a single, all-in-one network solution. As part of the Cisco Small Business product family, this Cisco UC320W Unified Communications is easy to set up and manage and flexible enough to grow and change with your business.
Here are some key features of Cisco UC320:
- Support for up to 24 phones – No License charges
- Inbuilt automated attendant with voicemail to email notification
- Integrated Wireless-N 802.11n wireless access point for voice and data
- Auto configuration of Cisco SPA 300 and SPA 500 Series IP Phones
- Interoperability with up to 12 public switched telephone network (PSTN) analog lines (FXO)
- Session Initiation Protocol (SIP) trunking support and SIP stack for clear, high-quality voice services
- 4-port Gigabit Ethernet (1000 Mbps) switch with VLAN support, to connect devices or expand
- Gigabit Ethernet WAN port that can be designated as the network edge
Additional features and benefits
The Cisco Unified Communications 320W provides great value for small business owners.
- Low cost: Cisco UC320 offers PBX system for both analog and digital system. You can use your old analog phones or digital phones like Cisco SPA500 phones.
- SIP Trunking: You can user SIP trunk to reduce your monthly charges and use analog lines as back up. This will ensure your PBX works even when internet connection is failed.
- Compact size: UC320 eliminates the need for big size PBX systems. Unit is very compact and can be installed on desktop or wall mounted.
- Comprehensive feature set: Innovative key system and small private branch exchange (PBX) capabilities
are available with easy-to-use Cisco SPA 300 and SPA 500 Series IP Phones, helping improve productivity.
- Investment protection: Businesses that are growing rapidly can use most components of the solution with
other Cisco Unified Communications solutions, providing industry-leading investment protection.
- Peace of mind: Cisco Unified Communications solutions deliver the solid reliability you expect from Cisco. All solution components have been rigorously tested to help ensure easy setup, interoperability, and
Incoming Call Showing Number of caller and The Number Dialed By the caller.
One of the key feature small business owners love about our VOIP PBX system is ability to route multiple phone numbers from various offices to one number. This allows them to reduce staff at multiple offices while providing better service to their customers.
This feature is also very popular with business owners running multiple businesses and/or websites but have the same staff answering calls for each business. For example, lets say you operate 5 websites and each website has a unique line of product, brand identity, domain name and toll free number. You can have all 5 toll free numbers come to your central office and route calls to your sales and support staff. Like i previously mention, by having all phone calls routed to same staff, businesses can reduce the number of employees needed to manage the business.
However, The challenge is how do you know which customer is calling what business ? This information is critical for sales staff to know so that they can properly greet the customer and help with correct product and/or service.
Solution is our Enhanced Caller ID solution. Enhanced called ID shows the number of caller just like any other phone but it also shows the number dialed by the customer. We can also convert the dialed number into name. For example, we can covert 510-709-4030 into www.fixtro.com. This feature quickly inform us that client is calling Fixtro.
Our system also shows this information in call log’s. You can quickly retrieve this information by checking missed call, dialed call or received call log.
Missed Call Showing Number of caller and The Number Dialed By the caller
Seeing the name is easier than the number. This feature allows you to see which office or domain customer is calling.
MAPPING PHONE NUMBER TO DOMAIN NAME OR OFFICE NAME
Many of customers also love getting their voicemail in email. Our enhanced called ID feature also shows all the details in email.
Let me know what do you think about this feature ? Can it help your grow you business ?
Sharmila Singh and Kern Singh, attorneys at The Singh Law Firm, provide estate-planning services to customers interested in protecting their future. Law Firm makes use of various tools to help their customers minimize estate taxes and built a better financial future. Their services include Business Succession Planning, Establishing Trust and preparing Wills.
Sharmila and Kern’s ability to solve financial problems and ease legal concerns awarded them with multiple referrals from their satisfied customers. Customer referrals along with infomercials on radio and web made The Singh Law Firm a popular destination for anyone seeking advise for protecting there hard earn money.
Rapid growth of Singh’s law practice significantly increased the number of phone calls receive and made during normal business hours. This provided an opportunity to upgrade their old analog system with next generation digital VOIP PBX System.
“I don’t want to miss a single phone call from my client,” said Karen Singh during initial interview with IT Project Manager from Fixtro.
After learning all the wants and needs of The Sigh Law Firm employees, Fixtro designed a custom VOIP solution that will ensure scalability and crystal clear connectivity.
Customer will be able to reach Kern, Sharmila and the office staff by simply dialing one office phone number. PBX system will automatically connect the customer to appropriate extension, cell phone or remote office. Office staff will be able to better serve their customer due to reduced hold time and efficent routing of calls.
The Singh Law Firm clients will find it even easier to get their appointment set up. No more holding on phone or engage signal. Simply call 510-742-9500 to set up your appointment today.
1. Research before Buying: Do research before signing with a VOIP PBX Provider. Number of VoIP service providers are growing by the day. With so many options in market, it can be quite confusing to pick a good service provider. I recommend you prefer a local vendor who can also support your IT infrastructure and meet with you on monthly or quarterly basis. This company will have a vested intreast to get you the most out of your VOIP budget. They will work hard to keep your phone systems out of trouble. You should also learn about the their products, services and capabilities before signing up.
2. Understand the Bandwidth Requirements: Start by making a good guess about how many simultaneous phone calls will made on average. On average a full T1 connection can handle up to 24 calls with no internet traffic. DLS and Cable internet speed can vary and should be considered carefully before implementing VOIP PBX Systems.
3. Static IP Address: Static IP address can make VOIP connection more stable. Check with your internet service provider to make sure they provide static, unchanging IP-address. Having a static address is a necessity for such tasks as peering two offices together for free phone calls, or having remote extensions for offsite employees.
4. Quality of Service (QoS): Start with the equipments you currently have in your infrastructure and make sure they support Quality of Service. QoS is the ability for a router or switch to prioritize certain types of traffic (like phone calls) over other traffic (like movie streaming).
5. Invest in Quality Equipments: Consult with good VOIP PBX system engineer for recommendation before buying your new equipment. They can guide you about the benefits and drawbacks for certain products. You will find two similar routers or switches can differ a lot when it comes down to real time performance required by VOIP PBX systems.
Tell me how your VOIP PBX is working out ? What would suggest to your friends when it comes to picking VOIP PBX System ?
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.
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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.