A network engineer receives a report about poor quality on an active call between the IP phone of user A over the WAN to the IP phone of user B. Using web access to the phone, the network engineer remotely checks call statistics such as jitter, network delay, and packet loss. Calculated packet loss is 3%, average jitter is 20 ms, network delay is 1 ms, and conceal seconds is 7.
What is the most likely problem with this call?
Packet loss greater than 1% is unacceptable for VoIP (Voice over IP) communications, as even small amounts of packet loss can lead to noticeable degradation in call quality. At 3% packet loss, the voice quality will likely be severely impacted, making it the most likely problem with the call.
With GETVPN, if a key server is configured to use multicast as the rekey transport mechanism, then under which of these conditions will the key server retransmit the rekey messages?
In GETVPN, the key server is responsible for distributing and updating the encryption keys to the group members. When using multicast as the rekey transport mechanism, the key server will retransmit the rekey messages if it does not receive the rekey acknowledgment from all group members. This ensures that all members of the group have received the updated keys to maintain secure communication. Therefore, the correct answer is that it only retransmits the rekey message when it does not receive the rekey acknowledgment from all group members.
Which four characteristics are associated with video? (Choose four.)
Video traffic characteristics include being greedy, delay-sensitive, drop-sensitive, and bursty. Video consumes as much bandwidth as is available (greedy), is sensitive to delays as they affect quality, reacts adversely to packet drops, and often transmits data in bursts.
What is the maximum amount of packet loss an engineer should allow for voice traffic on an IP network?
Voice traffic is extremely sensitive to packet loss, as even small amounts can significantly affect call quality. The industry standard best practice is to allow no more than 1 percent packet loss for voice traffic to maintain an acceptable quality of service. Higher levels of packet loss can result in poor call quality and intelligibility issues.
A Cisco IP phone is connected to a Cisco switch and is trying to obtain its network configuration. What is the first protocol that is used?
When a Cisco IP phone connects to a Cisco switch and tries to obtain its network configuration, the first protocol used is CDP (Cisco Discovery Protocol). CDP allows the IP phone to learn about its upstream switch and its VLAN assignments. Once these VLAN assignments are known, the phone then proceeds to use DHCP (Dynamic Host Configuration Protocol) to obtain an IP address and other network configuration details. Thus, the process starts with CDP.