- Posted by SoliCall
- On April 1, 2011
- 0 Comments
Many organizations today want to deploy / develop VoIP applications. The need for audio communication is required for many applications like customer support, internal communication between employees etc. VoIP is used as the underlying technology because it is the industry standard for voice communication, it can give high quality and it is usually a low cost solution.
Following is a list of parameters you should consider before deciding on the vendor/architecture for your VoIP application.
Sampling rate is measured in Hz. It is the number of times that the application is sampling the audio in a second. This parameter is crucial for audio quality. The minimum sampling rate that is used for voice is 8khz – i.e. 8000 samples per second. Sometimes this is also called narrowband. The narrowband is a legacy that gives basic audio quality. For a good audio quality you should at least use wideband. The wideband is using sampling rate of 16khz. It means that wideband is using twice the number of samples than narrowband. As a result the audio quality of wideband is much better than the quality that is achieved using narrowband. But this is not the end. As a reference, CD quality is 44.1khz. To achieve CD quality, high end VoIP applications, like Skype, are using sampling rate that is larger than 40khz.
Bandwidth is simply the amount of data that can be passed in the network in a specific period of time. Since VoIP is transferring the audio in the network it is crucial to make sure you have enough bandwidth to support it. Due to this reason the audio is usually compressed before it is sent on the network. The compression/decompression component is called codec. Bandwidth is usually not the first biggest problem for VoIP quality since audio requires relatively small bandwidth. For example if you are using wideband with the standard G.722 codec you do not need more than 48kbit/second. This is far less than the minimum requirement for streaming video.
There are several important standard protocols for VoIP. Some of them describe the signaling protocols (e.g. SIP & H.323) and others relates to the media and control protocols (e.g. RTP/RTPC). Almost all telephony vendors are fully supporting the industry standards so you can pick the best of bread solutions from different vendors. The consequences and limitations of using a solution that relies on proprietary protocols are obvious. Being bound to a single vendor has never been a good move…
Quality of Service (QoS)
The VoIP traffic is passed on the network together with the rest of the data. As a result the audio packets are fighting on the same network resources with the data packets. As a result you can get degradation in the audio quality. A solution for this common problem is to use QoS in the network that specifies that audio packets are getting precedence over data packets. In some cases it is even advised to use two different networks – one for audio and one for data.
Audio Quality Filtering
It is important that your VoIP solution will contain filters for improving audio quality. The most important filters are Acoustic Echo Cancellation (AEC), Noise Reduction (NR) and Automatic Gain Control (AGC).
If you plan on enabling conference calls, it is very important to understand how the audio mixing is being done. Good audio mixing algorithms have built-in mechanisms to improve audio quality, prevent echo etc. You should make sure you know and have control over this critical component.
How does your solution going to scale up with the increasing number of users without lowering the quality? What will happen if the number of participants in conference calls increases? You need to note that a solution that is good for a regular call might not work for conference calls and vice versa. Therefore, you might need a solution that has different scaling paths one for regular calls and one for conference calls.
Virtual Private Network
Running your VoIP packets on the public Internet might have quality implications on the audio. Depending on the changing load of the network with no QoS, your audio packets might suffer from time to time from delays, loss & jitter. One solution for this is to purchase a virtual private network (VPN) with guaranteed bandwidth & QoS. This technique can be used, for example, when you want to guarantee high audio quality between multiple sites in your organization.
Adaptive Multi-Rate Codec
If the VoIP application encounters a network congestion (e.g. when a specific user suffers from poor connection to the network) it should be able to temporarily lower the bandwidth that it is using. On expense of temporarily lowering the audio quality you increase the chances of supporting basic audio quality.
Sooner or later your VoIP network will have to communicate with other telephony networks. For example, you would want to enable to place/receive calls from mobile devices, land-line phones, other VoIP phones etc. You need to verify this option is available for you without compromising on quality. If you are using standard VoIP protocols you will have a variety of vendors/solutions to achieve high quality interoperability.