Added: 04/12/2016 • Written By: John Noblin
So you’ve decided to switch to a VoIP based provider—or maybe you converted years ago. Either way you know that VoIP systems have perks for businesses of all sizes, including cost savings, network integration, and higher productivity.
However, unlike standard telephony, VoIP sometimes has a tendency to drop the connection mid-way through a call for no apparent reason.
There are a few common reasons that this happens, and a few ways you can troubleshoot the issue:
Talk-off happens when your voice is improperly detected as a Dual Tone Multiple Frequencies (DTMF). Typically, these signals are triggered when you use the phone keypad.
A false trigger of the tone detector will not always cause a call to drop, but it is possible for a caller’s voice to be interpreted as a request to end the call. The remote server or PBX detector can misinterpret speech frequencies and drop a call or put it on an unintended “hold”. This tends to happen with women’s voices more than men.
Solution: To curtail talk-off, lower the gain on the phone’s microphone, or if the problem persists, reduce the sensitivity of the DTMF detection through your PBX admin controls.
Sometimes a connection fails and the failure is not detected immediately in a VoIP call. To prevent this, the Session Initiation Protocol (SIP) Timer periodically refreshes by sending repeated INVITE or UPDATE requests from one end-point to the other, typically in 10-15 minute intervals.
This allows both the user agents (signal sender/receiver) and proxies (signal route) to determine whether the SIP session is still active. If the expected message doesn’t arrive on time, then the user agent will send a BYE request and end the call.
It is easy to tell if this is happening if most VoIP calls timeout at the same duration (10-15 mins), whether someone is speaking or the line is silent.
Frequently, a malfunctioning SIP Timer is the result of small incompatibilities in endpoint devices from different manufactures.
Solution: To resolve SIP Timer issues, try adjusting the Min-SE settings on your device.
In some instances, a VoIP server may read no audio as a failed connection. By searching the media stream (RTP protocol), IP networks only transmit information when an audio signal is present.
Typically, there is enough ambient noise to keep the connection alive; however, muting a call can prompt an error. This can also happen if your device has a silence suppression or voice activity detection (VAD)—which helps save bandwidth, but stops sending audio when the volume falls below its limit.
You’ll know this is your problem when your call drops after a period of silence or extended use of the mute button.
Solution: Alter the silence suppression or VAD settings or dig around to see if your device has a specific configuration for this issue.
Session Initiation Protocol (SIP) sets a certain timeout period. In this process, the acknowledged (ACK) signals confirm that the callee’s device has received a final response to an INVITE request.
Typically, the call sets up, connects, and after 10 seconds the audio stops because the request failed to reach the intended destination after the timeout period elapsed.
Solution: This issue is a little more complicated and requires help from an expert to fix a bug in a SIP server. However, if you can find settings on your IP device related to NAT (Network Address Translation), you can tweak it to see if it makes a difference.
Many of today's commercial routers implement SIP ALG (Application-level gateway) as a default setting by the manufacturer. This feature is intended to prevent some problems caused by router firewalls by inspecting VoIP traffic and modifying it, if necessary.
While ALG can help solve NAT problems, in many cases it is implemented poorly and actually causes problems by corrupting information and making it unreadable. Ultimately, it can cause your incoming calls to fail.
Solution: If you experience this, check your router settings and turn SIP ALG off if it is enabled.
These issues can be managed or eliminated with TEC products such as VoIP Advantage and TEC Flex.
If your VoIP products need improvement, call TEC at (800) 832-2515. We’re experts at installing and managing VoIP Hosted Solutions for businesses.