VoIP (voice over IP) itself and Voiptime Cloud solution performance are both very sensitive to the underlying network and bandwidth. Inconsistency of the bandwidth can cause poor VoIP connection and, in such a way, problems with call quality.
VoIP call quality depends mostly on your Internet connection, local infrastructure, network settings, and headset.
Please use this short checklist to allocate the reason for the existing call quality issues.
Types of audio issues:
*Wi-Fi connection is not recommended for VoIP telephony because of its instability. We recommend the usage of wired internet connection only, with the minimal speed 100 kbit for each call. With the usage of WebRTC codecs this parameter can be decreased up to the 40-50 kbit for a call.
If none of the suggested decisions helped, please make the network diagnostics as is described below and attach results to the ticket.
Please run the test in the standard working situation (you shouldn’t close opened tabs and programs you use).
If you use Linux, run such command in the console:
mtr -s 1000 -r -c 1000 -o “LDRSNBAWVGJMXI” ts.voiptimecloud.com
|Packet loss||0% packet loss|
|Jitter||less than 20 ms jitter|
|Network latency||less than 100 ms latency to Voiptime Cloud data center. VoIP services are known to work even in higher latency conditions up to 150-200 milliseconds. However, this must be maintained consistently with no packet loss.
Internet connections from outside the United States are subject to higher latency, which can affect the Quality of Service (QoS).
|Bandwidth requirement||G711 Codec: 90 kbps symmetric/call
G722 Codec: 90 kbps symmetric/call
Please make sure you have 50% of your available bandwidth free to accommodate any spike in usage.
Always assume that at least 35% of your users are on call at any time. However, depending on your company’s use case, you may have a higher percentage.
|IP and RTP Traffic||Do not restrict SIP and RTP traffic across LANs and/or WANs when multiple customer sites exist.|
If you are interested in QoS and want to prioritize voice traffic, please contact our IT team and describe your current infrastructure.
QoS is highly complicated since it presupposes a lot of requirements both on our side and your WAN and LAN connections. To meet these requirements you should do the following: traffic marking, traffic prioritizing, etc. Should you be interested in fulfilling the above-mentioned requirements, we need to work on this together with our IT department.
|Port||TCP / UDP||Outbound||Description|
|443||TCP||Inbound / Outbound||app.voiptimecloud.com
Web application (https)
|7443||TCP||Inbound / Outbound||ts.voiptimecloud.com
Internal communication for WebRTC phone
Connection with the external VoIP carrier
To improve the call quality please follow the recommendations below:
Run the network test using PingPlotter and send us the results to check the situation with your current network settings.