お客様は初心者としても、弊社Avaya IX™ Calling Design Exam試験問題集の勉強方法やトレーニングガイドはあなたに適用され、Avaya IX™ Calling Design Exam認定試験に合格するのを助けます。

もしお客様は我々のAvaya IX™ Calling Design Exam試験問題集を購入すれば、ただほぼ20時間がかかるだけで、試験のレベルに達成することができます。それで、お客様の暇の短い時間をもって、我々のAvaya IX™ Calling Design Exam試験学習資料を勉強してから試験に参加できます。

デモをダウンロードする

我々のAvaya IX™ Calling Design Exam試験問題集は過去の試験データによって、すべてのエラーの問題が完全に削除し、改善します。それで、我々の問題集の正確性を高めます。20～30時間の学習で相応の効果を発揮することができ、効率的に試験に通過します。

三つのバージョン

我々会社のAvaya IX™ Calling Design Exam試験勉強資料は3種類のバージョンがあります。第一種はPDF版で、お客様は印刷してから、紙質の形式で勉強し、メモをできます。第二種はAvaya IX™ Calling Design Exam ソフト版で、真実の試験環境を模擬し作成されて、試験の雰囲気と流れを体験させることができます。第三種はオンライン版で、お客様はスマートとIPADなどの電子設備の上に使用されます。便利持ちなので、どこでもいつでも学習できます。

全額返済保証

当社31860X試験問題集をもって、簡単に試験に合格するのを助けますが、我々の31860X試験勉強資料を使用して合格しなかった場合に、あなたに全額返金することを約束します。私たちの唯一の目的は、あなたが簡単に試験に合格させるふことです。

Avaya IX™ Calling Design 認定 31860X 試験問題:

1. On an Avaya Aura 7.x deployment with shuffling turned on within Communication Manager (CM), which statement describes the media processing resources or Digital Signaling Processor (DSP) requirements for a SIP-SIP two-party call between endpoints using the same codec?

A) This call cannot shuffle, and two media-processing (DSP) channels are required for the entire call duration.
B) SIP-SIP two-party calls will use SIP Direct Media, so no media-processing (DSP) channels are required for this call.
C) SIP-SIP multi-party calls do not shuffle, and after setup use one media processing (DSP) channel for the entire call..
D) This call will shuffle after initial call set-up; i.e., one media-processing (DSP) channel is required for the first 20 seconds of the call, for each party.

2. A technician is adding 1100 users to an Avaya Aura solution. There is only bandwidth for 30kbps calls per user.
Which audio codec will provide the highest fidelity that the network will support?

A) H.264
B) G.729
C) G.722
D) G.711

3. You are designing a branch location that needs:
* 15 H-323 Endpoints
* 20 SIP Endpoints
* 1 DS-1 for PSTN trunking
* 2 Analog Ports
* Basic Survivability for all Non-SIP endpoints in event of WAN failure What is the least costly solution that meets all of these requirements?

A) G430 Media Gateway with S8300
B) G430 Media Gateway
C) G450 Media Gateway with S8300
D) G450 Media Gateway

4. If a customer has an Avaya Aura 7.0 and shuffling is turned on within Communication Manager, which statement describes the media processing resource (DSP) requirements for a two-party call originated statement describes the media processing resource (DSP) requirements for a two-party call originated by a SIP station and terminated by an .H323 IP Trunk being recoded?

A) SIP-SIP two-calls calls will use SIP Direct Media, so no media-processing channels are required for this call.
B) Multi-party (i.e. 3-party or greater) calls cannot shuffle, and require media-processing channels for the entire call.
C) This call will shuffle after initial call set-up; i.e., one media-processing (DSP) channel is required for the first 20 seconds of the call, for each party.
D) This call cannot shuffle, and one media-processing channel is required for the entire call.

5. You are recommending the WAN link requirement per branch for a multi-location customer. The customer estimates the branch Inter-office communication use rate over the WAN to be one trunk per 20 branch users, and there are 460 users at each branch. You assume that after all header compressions have been applied, there is a 30% protocol overhead, and the G.729 codec being used provides 7.2 kbps.
How much bandwidth is needed for voice traffic on the WAN link for each branch?

A) 215.3 Kbps
B) 115.9 Kbps
C) 165.6 Kbps
D) 187.2 Kbps

質問と回答：