Don’t Let Your Video Get Lost In Public

February 10, 2014, by Lisa Avvocato in Cloud Services, Video Conferencing

The proliferation of desktop and mobile video solutions, along with WebRTC, has allowed participants to join a video call virtually anywhere there is an internet connection. However, a poor internet connection can destroy a video conference. Here are a few things you need to know when joining a video call over the internet:

Download Speed:
Download speed is the amount of bandwidth people have coming to their computer from the Internet. Think of a road coming toward an office; the more lanes it has the more traffic the road can handle. Similarly, the more downstream bandwidth people have the more internet traffic they can accept. For a point-to-point business quality video call it is recommended to a minimum download bandwidth of 384Kbps. For each additional call participant an additional 384Kbs is recommended. For example, a 4-way call will need 1.5Mbps + 20% for overhead. For High Resolution (HD) video conferencing, a minimum of 1Mbps (+20%) download speed is recommended.

Upload Speed:
Upload speed is the amount of bandwidth people have going from their computer to the Internet. This is the road going away from an office.  Again, the more upload bandwidth one has, the wider the road is and the more traffic people can send. The upload requirements remain the same as the download requirements regardless of the amount of participants on the call.

Latency (Delay):
Latency is the amount of time it takes for the traffic sent to reach its destination. Using the previous analogy, even if there is a wide road going to and from the office, if a car is moving slowly on the road it will take a lot longer to get where it is going. If you notice it is taking a long time for your co-worker to respond on a video call or that you are talking over each other it is most likely being caused by high latency. Latency problems are often caused by network congestion; if you experience problems try ending the video call and starting it again. It is recommended that latency be below 250ms.

Jitter:
Jitter is the time difference it takes data packets to reach their destination and is usually caused by congestion in the network. This is akin to getting off of work and hitting the evening rush hour. Due to the congestion and high volume of drivers hitting the road at the same time it may take longer to reach your final destination.  Jitter causes packets to arrive at their destination with different timing and possibly in a different order than they were sent (spoken). Some arrive faster than they should while some arrive slower than they should. Low jitter, or a few packets off causing a slight flicker or flash, can be frustrating but tolerable.  High jitter on the other hand can make video nearly impossible to use as the image can be completely distorted. It is recommended that jitter be below 30ms.

Packet Loss:
Packet loss is when one or more of the data packets fail to reach their destination and is also caused by congestion on the network. Essentially, some of the packets are dropped by network routers or switches that become congested (lost packets), or they are discarded by the jitter buffer (discarded packets). This is similar to an audio call breaking up where you miss every few words and cannot understand much of the conversation.

Test Your Network:
There are a number of ways to test your network connection both for quality as well as any firewall/security restrictions. Check out IVCi’s new Cloud Video Experience Video Network Assessment test to see how well video is expected to perform on your network. Click here to star the test.

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