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NEW QUESTION: 1
In der Bestellanwendung führt ein Benutzer den Bericht "Bestelldetails" aus. Die Anforderungsseite des Berichts wird nur mit Planungs- und E-Mail-Informationen angezeigt. Warum fehlen die Parameterinformationen auf der Anforderungsseite?
A. Der Bericht wird mit der aktuellen Abfrage der Anwendung ausgeführt.
B. Der Bericht wird für die gesamte Datenbank ausgeführt.
C. Der Benutzer hat keine Sicherheitsrechte für die Parameter.
D. Parameter sind nicht nur für die Einkaufsanwendungen verfügbar.
Answer: C
NEW QUESTION: 2
別紙をご参照ください。
SW2はこのVTPドメイン内の他のスイッチとどのように相互作用しますか？
A. トランクポート上のネットワーク上のVTPクライアントからのVTPアップデートを送信して処理します
B. トランクポートで受信したVTPアドバタイズのみを転送します。
C. アクセスポート上のネットワーク上のVTPクライアントからのVTPアップデートを処理します。
D. すべてのVTPサーバーから更新を受信し、ローカルに構成されたすべてのVLANをすべてのトランクポートに転送します
Answer: B
NEW QUESTION: 3
ユーザーがEBSボリュームのスナップショットを作成しました。スナップショットに関して不可能な使用方法は次のうちどれですか？
A. ボリュームのサイズを大きくします
B. インスタンスを起動します
C. 1つのAZから別のAZへのボリュームのミラーリング
D. ボリュームサイズを小さくする
Answer: D
Explanation:
EBSスナップショットは、ボリュームのポイントインタイムバックアップです。ボリュームを1つのAZから別のAZに移動するか、新しいインスタンスを起動すると便利です。ユーザーはボリュームのサイズを増やすことができますが、元のスナップショットのサイズより小さくすることはできません。
http://docs.aws.amazon.com/AWSEC2/latest/UserGuide/EBSSnapshots.html
NEW QUESTION: 4
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. 480 bps
B. 240 bps
C. 20 bps
D. 150 bps
E. 640 bps
F. 40 bps
Answer: D
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.