on VoIP and SIP Trunking

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Basic Guide to VoIP Troubleshooting

Diagram on the left showed the flow of VoIP in a Network and the Area of Problem Section.

Echo Problem

Echoing is one of the most common problems and is often the easiest to fix.

Symptoms – One or both parties hear one (or both) voices repeated, often faintly, with a slight delay.

Related Equipment – #1: The handsets or softphones; #10: Other party’s phone.

Cause – Echo is usually caused by ‘crosstalk. That’s when a phone’s microphone picks up the sounds from the phone’s earpiece because the volume is too high.

Resolutions – Have both parties turn down their handset volume.If using a softphone without a headset, turn down the volume, or better yet use a headset. This will keep the mic from picking up sound from the speaker.

Choppiness Problem

People usually say the call is “breaking up” when they hear choppy audio.

Symptoms – It sounds like words, or parts of words, are missing.

Related Equipment – #2: The router or switch on the LAN; #3: The firewall on the office’s LAN; #8: WAN; #9: WAN transmission towers.

Cause – Choppiness is caused by ‘packet loss’ which results from the ‘high jitter’ of an unstable connection.

Resolutions – Ensure the firewall is properly configured to allow for and prioritize VoIP traffic. If the office uses the same LAN for computers and VoIP, then ensure that no computers on the LAN are using more bandwidth than normal. Streaming video sites (such as Hulu and Netflix) and video conferencing are two common culprits.

You may need to perform some tests on the two network variables which most affect VoIP call quality, latency and jitter. These, along with packet loss, are the main variables by which network and connection quality are measured. They are defined as:

Latency: Latency is the amount of time it takes for packets to arrive, and it is measured in milliseconds (ms, meaning thousandths of a second). The maximum one-way latency (from one caller to the other) that can still allow for high-quality VoIP calls is around 150 ms. Latency is measured by doing a ping test.

Jitter: Jitter is the variation in the latency of arriving packets. Remember that packets often arrive having taken different routes, so some arrive more quickly than others. Jitter is measured by doing a traceroute test.

Packet loss: If some packets take too long to arrive, the others do not wait. The voice audio will be reassembled without the late or missing packets, resulting in packet loss.

Try an online connection speed test to verify that upload and download speeds are consistent with what your ISP is providing.

If necessary, consider increasing bandwidth from the ISP or switching to a Dedicated Voice broadband SIP Trunk. 

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