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We all know that businesses never run as smoothly as we’d like them to. We try relentlessly to keep our business(es) running like a well-oiled machine. Unfortunately, every now and again there are certain components of that machine that can occasionally break and need some repair. As you grow your business you should begin to treat it as you would with that sporty two-door Benz parked in your garage. Bring it in for an inspection and catch any issues that may arise beforehand that could prevent a potentially catastrophic risk, break-down, or accident later.
When starting your own VoIP business, it’s always important to survey the hazardous landscape. Little do most know, it would be relatively easy to avoid the traps that so many of us have fallen into. Thankfully we’ve taken it upon ourselves to equip you with the knowledge you need to avoid these problems. In this post we’ll cover the top 10 technical pitfalls that you should be aware before your VoIP journey begins.
1. Post-Dial Delay
Post-dial delay (PDD) is the delay experienced between the time you dial a number and when you receive an audible signal. It is typically encountered when placing an outbound call on both SIP trunks and/or Hosted phone systems. Why does post-dial delay become a problem? It varies depending on the terminating destination’s response. It may result in a few seconds of perceived missed ringing to no ringing at all with just hearing the called party answering. SIP signaling relies on the called (termination) party to tell the receiving (originating) a message instructing what kind of signal to relay. Is there a remedy? As the SIP protocol operates primarily on industry recommendations there is currently no way to force carriers to always provide the messaging that is required. As SIP evolves this may be an issue that is left behind. However, for the time being, it is a fact of life when dealing with SIP.
2. One-Way Audio
One-way audio is when one side or one party can hear the other, but not the reverse. Typically this is indicative of something stopping either the outbound or inbound from reaching the receiving party. Finding the cause will require simplification of the connection by eliminating the use of some equipment then testing the call. Simply check your equipment to be sure that you have audio in both the earpiece and microphone.
Try using a recording technology such as Windows Sound Recorder. When your equipment is working properly, you can then simplify the connection. Do this by connecting the VoIP connection directly to the modem device. Proceed by making a test call. Lastly, reconfigure your LAN network. This is typically done by eliminating a second layer of NAT from your network design.
3. PBX/Switch Configuration
It’s probably obvious that you want your SIP and RTP traffic to flow properly. So make sure your PBX Switch is properly configured. In most cases, when using a hosted PBX your features are hosted by a service provider at their location. You would then connect via IP to the provider. An IP PBX consists of one or more SIP phones, an IP PBX and optionally a VoIP gateway, to connect to existing PSTN lines. Make sure your SIP trunks are set up properly via your provider and calls should be running smoothly.
4. Audio Quality
You’ll find two common issues with call quality on VoIP calls: jitter and latency. Jitter is a common problem of the connectionless networks or packet switched networks. Since the information is divided into packets, it arrives at their intended destination in a different order than they were originally sent and the result is a call with poor or scrambled audio. The solution? Use jitter buffers. Jitter buffers temporarily store arriving packets in order to minimize delay variations. VoIP delay or latency is characterized as the amount of time it takes for speech to exit the speaker’s mouth and reach the listener’s ear. Latency sounds like an echo. The solution? Policy-based network management, bandwidth reservation, type of service, class of service, and Multi-Protocol Label Switching (MPLS) are all widely used techniques for prioritizing VoIP traffic.
5. Network Security
Having poor network security will put you out of business, in debt, and could possibly lead to huge legal issues. So be secure! Fraud protection can save your life. If you don’t have it, DDoS (distributed denial of service) attacks won’t knock at the door, they’ll just let themselves right in. Ask your provider what best practices for resolving interception of calls, DDoS attacks, theft of service, and exfiltration of data you should implement to prevent them from ever happening.
6. Automation For End-Users
Choice, automation, and control are three key components of an efficient end-user experience. You want your end-users to have the capabilities to conduct some of the heavy lifting for you such as adding, editing, and deleting their services. The last thing you want to deal with in business is time-killing pains that prevent you from growing. Utilizing an end-user portal gives your customers a branded portal where they can change Caller ID, names, and 411 information. Tie that together with a helpful set of APIs to guide them and your customers will be clear on any of the routines, protocols, and tools they’ll utilize through the entire process.
7. Redundancy
Implementing redundancy at the VoIP and network infrastructure levels is an integral part of VoIP communication, ensuring call and business continuity. Having call continuity and system redundancy features are crucial for providing your customers with the peace of mind of no calls ever being lost. These features also automatically forward calls in real time to designated mobile phone numbers if your internet service fails. Many systems also incorporate disaster routing. If the system cannot connect to the user’s private telephone network within four seconds, the call is automatically routed to one of up to three other systems specified by the user.
8. DTMF (Dual Tone Multi-Frequency)
DTMF (Dual Tone Multi-frequency) are signals/tones that are sent when you press a telephone’s touch keys. (For example, if you call VoIP Innovations and press “1” for Sales or “2” for Provisioning.) What are the problems that may arise with DTMF? Packet loss–The inband method of transmitting DTMF suffers the occasional loss of packets and overall connection. Troubleshooting DTMF issues are hit and miss and may be as simple as using a different DTMF setting and retrying. But, before making any changes with settings make sure that the issues that you are experiencing are not related to packet loss. Packet loss can create havoc with VoIP connections and in relation DTMF tones.
9. Faxing Over VoIP
We all know faxing has been around for years. Most of the protocols for faxing were written with the intent of sending those signals over traditional phone circuits using sounds. These sounds were then turned back into data by the receiving fax machine, which expects a consistent data transmission, without any loss.
In VoIP, voice is first converted into packets and then they are sent over the connections that make up our vast internet. Some packets may become discarded, but the end result is a clear and understandable conversation. To avoid any information or packet loss, slow the transmission rate down and allow your fax machine to continue to receive a transmission, allowing a few bits of data to be lost, but also allowing the machine to become more consistent.
10. Call Completion (Termination)
The importance of having a trustworthy and comprehensive A-Z termination service and provider is crucial. VoIP termination is used to refer to the features that are used for routing telephone calls from one provider to the next until the call has been routed to the last telephone company and has been received by the recipient. If your termination provider is faulty, calls will be lost and so will your customers. When looking for a termination provider, it is important that you choose a service provider who is reputable, trustworthy, and is aware of your business communication requirements. It is also very important to keep in mind the size and scale of your business before choosing a VoIP termination provider.
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