Common Causes of Poor Voice Quality and Video Pixelization

A.

Poor voice quality and video pixelization in a video call can be caused by a variety of factors. Here's a detailed explanation of the possible causes:

A. The QoS is configured incorrectly: Quality of Service (QoS) is a set of technologies that ensure that network traffic is prioritized according to its type and importance. If QoS is not configured correctly, voice and video traffic may not be prioritized over other types of traffic, such as data traffic. This can result in poor voice quality and video pixelization due to packet loss, delay, and jitter.

B. A firewall is blocking the RTP ports: Real-time Transport Protocol (RTP) is the protocol used to transmit voice and video traffic over IP networks. If a firewall is blocking the RTP ports, voice and video traffic may not be able to reach their destination. This can result in poor voice quality and video pixelization due to dropped packets and packet retransmissions.

C. Cisco Unified Communications Manager is configured to use D. 711 instead of E. 729: G.711 and G.729 are two popular codecs used to encode voice traffic in Cisco Unified Communications Manager (CUCM). G.711 provides higher voice quality but requires more bandwidth than G.729. If CUCM is configured to use G.711 instead of G.729, voice traffic may consume more bandwidth than available, resulting in poor voice quality and video pixelization.

D. 1 Gbps network ports are used instead of 100 Mbps network ports: If a video call is using more bandwidth than the available network capacity, it can result in poor voice quality and video pixelization. If 1 Gbps network ports are used instead of 100 Mbps network ports, the available bandwidth may be insufficient to handle the video call traffic. This can result in dropped packets and packet retransmissions, leading to poor voice quality and video pixelization.

In conclusion, to ensure high-quality voice and video calls, it's important to configure QoS correctly, ensure that firewalls are not blocking RTP ports, use the appropriate codec for voice traffic, and make sure that the network capacity is sufficient to handle the traffic.