Basic VoIP/SIP Considering Factors
In today’s day & age, we expect that everything runs optimally at all times, specifically for essential services like our internet and phone service.
Your current provider may cause some VoIP issues, but what about your network? Your bandwidth, configuration, etc.?
Why is bandwidth so crucial in a VoIP environment?
What is Bandwidth?
The maximum data you can transfer through your internet connection during a specific time is called bandwidth.
Bandwidth is different from your data transfer speed because it is the capacity of your network communication. The most common way of measuring bits per second is Kbps or Mbps.
VoIP or Voice over IP uses a series of codecs to decompress and compress your voice data. A codec enables you to decode a signal or digital data stream. A secure internet connection is required for a VoIP solution. If you don’t have a stable internet solution, you will experience call quality issues.
How Is VoIP Quality Affected?
Your internet speed can impact the quality of your VoIP system. A high-speed internet connection is essential for clear VoIP resulting in high-quality calls through your VoIP phone system.
What you may not realize is the importance of your network bandwidth. If you do not have enough bandwidth, your VoIP will not be usable. Even if your VoIP is clear, it can be impacted due to traffic spikes within your network.
The result can be a noticeable drop in sound quality, often causing static. There is also the possibility your calls may drop altogether.
If you are part of a shared network, your VoIP voice quality can be affected.
When the load of a network is too heavy, your voice quality can decrease. Your best option is to give voice packets a much higher priority.
You need to take into consideration your call setup rate, success ratio, and setup times. The protocol of your network will also affect your voice calls.
Slow Internet Connection
The quality of your VoIP calls will be significantly affected by your internet connection speed.
You will be unable to receive good voice quality if you use either a shared network or a dial-up connection.
The majority of broadband connections work exceptionally well with the technology necessary for VoIP.
It would be best to have an audio codec when using VoIP technology to compress audio signals. It is also essential to decompress audio signals on your receiver end.
The majority of audio codes currently available are relatively standard. Keep in mind that your VoIP provider provides you with a proprietary code. Some of the most well-known codes include G.711u and G.729a.
Your audio codec significantly impacts your VoIP voice quality. Your audio codec compression factor determines the voice data size you can transmit through your network.
Very little data will need to be transmitted with a high compression rate, but your voice quality will deteriorate. This is why most audio codecs offer a compression rate of 8Kpbs, 6.4 Kbps, or 5.3 Kbps.
You will only use these bitrates for voice. Protocol overheads have also been added.
It means the rate of your actual bit will always be higher.
Because of your audio codec, you will have a digitizing delay between all of your transmissions. The algorithm first compresses your data before being sent to your receiver. Once this is accomplished, your data is decompressed at your receiver end.
Additional CPU resources are necessary when an audio codec requires a complex algorithm.
The voice quality you experience on your calls can be affected by this algorithm.
Voice Packet Loss
Do not be surprised when some voice packets are lost in your network because this happens automatically.
There are numerous reasons, such as excessive collisions, media errors, and overloaded links.
You can recover lost audio packets with many different protocols, including TCP.
You will be unable to recover lost packets if you use specific protocols such as UDP.
If there is a lengthy distance between the sender and receiver, the result is often latency. Latency usually occurs when network conditions are weak.
When the distance from the receiver increases, so does the latency. Latency is also dependent on how many routers your packet must visit before reaching the final destination.
Your delay time can also increase due to audio codecs. Time is necessary for the execution due to the complexity of the algorithms.
You can be assured of clear calls when your network latency is under 150 ms and remains stable. As long as there will be little to no variation with your latency, calls will remain crystal clear.
VoIP IP media is converted to analog or digital signals.
Your network will introduce an echo. The two different echo are Hybrid echo and Acoustic echo.
Containing and monitoring an echo is extremely difficult. Find a provider offering echo cancellation modules. This is an excellent method to decrease the echo you have in your network.
If your equipment is outdated, it will impact your VoIP calls. Proper routers, firewalls, and cable modems are required.
Your voice quality can also be affected by the frequency of your phone calls. You will receive better voice quality by using lower frequency phones.
The router you decide to use must support a firewall, audio codecs, and echo cancellation.
Most smaller businesses use an internet connection for both data and voice. This means the router must give priority to VoIP traffic. Otherwise, your call quality can degrade when just one user downloads a large file. You can solve this issue by purchasing a relatively cheap specialized VoIP router.
An issue may also occur that, unfortunately, can be out of both parties’ control due to an intermittent problem with your internet provider.
As long as your network has been configured with Quality Of Service prioritizing your VoIP communication, the correct codec and good bandwidth, your communication will remain crystal clear while remaining cheaper than any PSTN provider.