The most common reasons for telephone issues may include:
1. Lack of power and internet connection.
2. Poor signal quality when connected via Wi-Fi.
Cable network connection provides higher and more reliable quality. While using Wi-Fi, difficulties may arise with signal strength, line congestion, or the router. All these factors can adversely affect communication quality.
To troubleshoot, we recommend switching to a different network and testing the SIP account functionality (for example, by switching to mobile internet on a mobile device or sharing mobile internet for a desktop device).
3. High load on the CPU and memory usage of your device
You can check the CPU’s (central processing unit) load and your computer's memory usage:
for Windows use hotkeys Ctrl + Shift + Esc
for MacOS: Activity Monitor => Window => CPU Usage.
It's not possible to define the state that is considered normal, but if you observe that the CPU is loaded at 90%, we recommend you close some tabs or programs.
4. Temporary blocking of the IP address due to incorrect registration data (username or password) or simultaneous registration of the SIP account on multiple devices
We recommend verifying the accuracy of the entered data (making sure there are no extra spaces at the beginning or end).
Additionally, please check whether the SIP account is not registered on multiple devices simultaneously. For proper operation, you should authorise the SIP account on only one device in one program, extension, SIP phone, or PBX at a time.
5. Headset malfunction issue
The microphone or speaker of the headset may be faulty. Ensure you check the audio settings on your device or in your browser, as you may have chosen the external device microphone over the headset microphone.
It is essential to check whether the microphone on the headset is functioning. For example, you can test the headset's performance with another device and try to use a different headset.
Furthermore, check the sound settings in the browser (if using a browser extension) or the device settings (if using a softphone application).
Communication issues and possible causes of their occurrence
1. Calls end, or there are beeps, but the operator/manager does not receive the call.
We recommend checking whether a call forwarding scheme is assigned to the numbers in the project. If it is set, it is necessary to verify its settings, specifically:
Ensure that the phone number in the scheme is correctly specified (should be in the international format, e.g., +44XXXXXXXXX) or use the SIP account login.
Verify that the time and days of the week are configured correctly (the scheme should cover all weekdays and times of the day).
2. Calls are not reaching the SIP accounts.
We recommend verifying the successful registration of SIP accounts, checking if they are online on both the device and in the user account:
If the SIP accounts have a status “online” but calls are not reaching them, we suggest checking for any error messages displayed in the program (for example, STUN server error).
If you see the STUN server error, we recommend trying to modify its settings: disable it, use the program's default STUN server, or specify our STUN server.
Additionally, to test the functionality of the SIP account, we recommend making an outgoing call to a phone number. If the call fails, the issue may be related to the registration of the SIP account or the device's functionality.
However, if the outgoing call is successful, we recommend checking the background operation settings. You should enable them for mobile device applications.
If the SIP account is authorised in the PBX, we advise reviewing the internal call forwarding settings in the PBX.
3. Sound Delay
More often than not, sound delays occur for two reasons: network issues or processing delays by the device.
It occurs when you connect to the internet not directly through a cable but via Wi-Fi.
Additionally, if the router is overloaded, the data for IP telephony cannot "breakthrough" and gets delayed.
In the first case, Wi-Fi may lack the capacity to ensure сontinuous data transfer. Cable internet provides higher and more reliable communication quality.
In the other case, websites will still load quickly during page surfing, as data may load in "fragments," which is unacceptable for telephony.
Telephony uses streaming, and sound should be transmitted continuously. SIP telephony doesn't require a powerful internet connection. But it should have a specific designation for operator conversations.
When you talk to someone using SIP telephony, your voice is initially encoded and then decoded at the "output" after transmission over the network.
Usually, it happens almost instantly and is unnoticeable to the person talking. However, if you make a call on a computer with an overloaded CRU, a processing delay occurs because the device can’t process tasks promptly.
To address this, we recommend:
Close unused tabs in your browser and background programs.
To rule out device issues, set up the SIP account on another computer and try making a call from there.
Configure traffic prioritising on your router— more details are in our blog article "QoS settings that will increase the quality of IP telephony."
Check how the connection works by connecting through another provider. For example, take your laptop home or to another office and set up the connection there or use mobile internet.
If the problem persists only in your network, we recommend contacting your provider to resolve it.
4. One-sided audio
If your interlocutor cannot hear you or vice versa, the issue may be related to your network settings.
The cause could be NAT (Network Address Translation) technologies. It’s commonly employed by most service providers and in home or office networks. NAT is designed to address the shortage of IP addresses and ensure the security of local networks connected to the internet.
To resolve this, we recommend:
Configure a STUN server for your softphone application or SIP phone. Our STUN server's address and port are stun.ringostat.com:3479.
Enable SIP telephony support on your router. This feature is present on most modern devices. You can find it in the online instructions.
If the previous recommendations do not resolve the issue, port forwarding for SIP telephony ports may be required. In this case, we recommend you seek assistance from a system administrator or a network configuration specialist.
Example of port forwarding:
The system administrator statically associates the IP address, such as 192.168.0.1, with the Manager user. It implies that the Manager's device will consistently be assigned the specified local IP.
Next, the system administrator opens port 5060 for UDP traffic for the specified IP address.
After that, the system administrator opens the port range 10000-20000 for RTP traffic for the specified IP address.
Security configuration step: The system administrator establishes a rule allowing connections for the configured ports only from the IP addresses of Ringostat telephony servers (you can obtain the server address hosting your project by contacting the technical support service).
5. Echo and low-quality sound during calls
Poor headset quality.
You or your interlocutor hold the handset too far away or have activated loudspeaker mode.
To resolve this, we recommend:
Use a high-quality headset, such as a USB headset from Jabra (Jabra Evolve 20 MS) or Sennheiser.
If you are using a non-USB headset, ensure you install the latest audio card drivers provided by the manufacturer.
If calling from a laptop, ensure you are not using its external microphone instead of the headset microphone. You can check this in the device's sound settings.
Use standard codecs supported by most telecom operators. Choose only the 711-A and 711-U codecs in the device settings.
Also, we recommend conducting tests to check your device's settings if you are using softphone applications on a computer or laptop, for example, Test Voximplant.
This test will allow you to check the following parameters:
Connection quality - speech clarity, signal loss.