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NEW QUESTION: 1
How do you enable notifications on workflows?
A. Configure a security group as supervisor in the workflow properties.
B. Enable the "Workflow notifications" checkbox under Administration > Application Settings > Workflows.
C. Configure a security group and a notification template that should be used when the workflow is in error.
D. Select the "Workflow status error" checkbox under Notifications in the workflow properties.
Answer: A
NEW QUESTION: 2
ホストプロファイルは、ホストプロファイルに接続されているすべてのホストで緊急に変更する必要がある新しいパスワードで更新されます。
管理者は、パスワードの変更を完了するためにどのオプションを選択しますか？ （最良の答えを選択する。）
A. ホストプロファイルのコンプライアンスを確認する
B. 修正
C. ホストから設定をコピーする
D. ホストのカスタマイズを編集する
Answer: B
NEW QUESTION: 3
To determine the bandwidth requirement for each VoIP call, not including Layer 2 overhead, how much bandwidth per call should be added to account for the voice signaling traffic?
A. 150 bps
B. 40 bps
C. 240 bps
D. 640 bps
E. 480 bps
F. 20 bps
Answer: A
Explanation:
Voice quality is directly affected by all three QoS quality factors such as loss, delay, and delay variation. Loss causes voice clipping and skips. Industry standard codec algorithms can correct for up to 30 ms of lost voice. Cisco Voice over IP (VoIP) technology uses 20 ms samples of voice payload per VoIP packet. Only a single Real Time Transport (RTP) packet could be lost at any given time. If two successive voice packets are lost, the 30 ms correctable window is exceeded and voice quality begins to degrade. Delay can cause voice quality degradation if it is above 200 ms. If the end-to-end voice delay becomes too long, the conversation sounds as if two parties are talking over a satellite link or a CB radio. The ITU standard for VoIP, G.114, states that a 150 ms one-way delay budget is acceptable for high voice quality. With respect to delay variation, there are adaptive jitter buffers within IP Telephony devices. These buffers can usually compensate for 20 to 50 ms of jitter.
