The VoIP-based phone system is one of the most popular telecommunications systems that are being used worldwide by several users. VoIP or Voice over Internet Protocol technology is playing a vital role in the effective communication of businesses. As compared to traditional communication systems, a VoIP network-based phone system is much better in terms of features, functionalities, as well as costs.
Benefits of VoIP phone system
VoIP technology allows users to make calls to any local or international number using the internet. The best thing about this system is that the users will be charged at local rates along with the usual internet price. The benefits can be summarized as:
- Cost-effective solution
- Better accessibility
- Highly scalable
- Advanced features
- Clear voice quality
- More flexible
Though there are several benefits of using a VoIP network-based phone system, there are few disadvantages as well. One such disadvantage is dropped VoIP calls that are mostly faced by the users. They experience a sudden call-off of a few seconds, usually about 30 seconds while talking over the phone. Generally, this issue occurs at the time of making outbound calls on high-volume networks.
Key reasons for VoIP call drop and their solution
There can be several reasons behind VoIP calls drop, some of them are mentioned below along with ways to troubleshoot those issues, have a look:
- Talk-Off: Talk-off occurs when the caller’s voice is inappropriately detected as a Dual Tone Multiple Frequencies (DTMF). Usually, these frequencies or signals are generated while using a phone keypad. It may be possible that the caller’s voice is interpreted as a request to end the call. Besides, the remote server or PBX detector can misunderstand speech frequencies, thus dropping a call or putting it on hold.
In order to resolve this issue, users can reduce the sensitivity of the DTMF detection via their PBX admin controls.
- Out of Order SIP Timers: Many times it happens that while making the VoIP calls connection fails. Sometimes, the failure is not recognizable right away in a VoIP call. To overcome this issue, a Session Initiation Protocol (SIP) timer is used that refreshes the session from time to time by sending repeated invite or update requests from one end-point to the other, usually in the interval of 10-15 minutes. This allows both the user agents (sender/receiver) and proxies (signal route) to confirm whether the SIP session is still active. If the expected message doesn’t turn up on time, then the user agent will end the call. Often, an out-of-order SIP Timer is the result of small incompatibilities in endpoint devices.
In order to resolve SIP timer issues, users can adjust the Min-SE settings on their devices.
- RTP Silence: In some cases, it may be possible that a VoIP server reads no audio as a failed connection. By searching the media stream (RTP protocol) IP networks send information only when an audio signal is there. Though, there can be some noise to keep the connection alive, muting a call can prompt an error. It may also happen that the device comprises a voice activity detection (VAD) feature that can save bandwidth but restrict the transmission of audio when the volume falls below its limit.
To resolve this issue, users can change the silence suppression or VAD settings.
- Incorrect routing of SIP ACK Signals: Session Initiation Protocol (SIP) usually defines a particular timeout period. During this period, the acknowledged (ACK) signals authenticate whether the caller’s device has received a final response to an Invite request. After the call sets up, it gets connected and suddenly after a few seconds, the audio stops working as the request fails to reach the anticipated target after the timeout period is completed.
To resolve this issue, the users can take the help of a professional to fix the bug in a SIP server. They can also try to find settings related to NAT (Network Address Translation) on their IP device and can alter it if it can work.
Router SIP ALG intervention: Several commercial routers these days employ SIP ALG (Application-level gateway) as a default setting by the manufacturer. It is a feature that assists users in eliminating certain issues caused by router firewalls. It helps to examine VoIP traffic and modify it if required. Though ALG helps to deal with NAT-related issues, if it is implemented badly, it can corrupt and make information unreadable. This may result in VoIP call drop.
To resolve this issue, users must check their router settings and turn off the SIP ALG, in case it is enabled.
VoIP is undoubtedly the best alternative for today’s businesses as compared to traditional copper-wire telephone systems. It not only offers bandwidth efficiencies but is also the most affordable option for setting up a flawless communication system. Its benefits outweigh its disadvantages. Those who are facing issues like Dropped VoIP Calls must go through the above-mentioned solutions to overcome some of the major VoIP call drop issues. By eliminating these kinds of issues, businesses including SMBs and large enterprises can enjoy flourishing growth in the form of better communication systems and enhanced customer experience.