Here you will find answers to Voice Over IP Questions
Which type of delay can lead to jitter in a voice network?
A. Propagation delay
B. Serialization delay
C. CODEC delay
D. Queuing delay
Jitter is the variation in the arrival of voice packets. For example, the first voice packet of a conversation might take 50 ms to reach a destination while the second voice packet might take 60 ms. There is 10 ms of delay variation (jitter) between these packets. The varying arrival time of the packets can cause gaps in the re-creation and playback of the voice signal. These gaps are undesirable and annoy the listener. For example, if the speaker says “Enjoy your life” then the listener will hear “Ennnnnnjoy yooooour liiiiiiife”.
Queuing delay (how long a packet waits in a router’s interface queue) is variable because it depends on how many packets are currently in the queue. Therefore queuing delay is the main reason leading to jitter in a VoIP network.
Other delays (propagation, serialization, CODEC) are fixed and predictable delays.
+ Propagation: The time it takes a packet to traverse a link.
+ Serialization: The insertion of bits onto a link.
+ CODEC: The time for translating the audio signal into a digital signal.
IP phone A places a call to IP phone B. How many RTP streams are required for the call to be successfully completed?
Notice that RTP streams are one-way. If you are having a two-way conversation, the devices will establish dual RTP streams, one in each direction
Refer to the exhibit.
The Acme Corporation needs assistance in configuring their PSTN voice gateway. Which two dial peers will correctly route calls to emergency services? (Choose two)
A. dial-peer voice 1 pots
B. dial-peer voice 911 pots
C. dial-peer voice 9911 pots
D. dial-peer voice 2 pots
E. dial-peer voice 1 pots
F. dial-peer voice 2 pots
Answer: E F
The first time I read this question, I think the Local PSAP (Public Service Answering Point) can accept both 911 and 9911 but it is not true. The Local PSAP only accept 911 so the duty of the administrator is to configure the gateway in order to support both 911 and 9911 numbers. To do this, we need two dial-peers, one for 911 and another for 9911.
But keep in mind that our outgoing dial-peer (port FXO 1/0/0) is a POTS dial-peer so the matched digits of this dial-peer will get stripped so we need to use the forward-digits all or forward-digits 3 (for 911 pattern) or prefix 911 (for 9911 pattern) to keep the called number. Therefore only E and F are correct.
Approximately what percentage of voice packets can be dropped before voice quality becomes poor?
A. 1 to 2%
C. 5 to 10%
D. Less than or equal to 1%
Packet loss describes an error condition in which data packets appear to be transmitted correctly at one end of a connection, but never arrive at the other. This might be because:
- network conditions are poor and the packet became damaged in transit
- The packet was deliberately dropped at a router because of internet congestion.
The 1% threshold is just an estimate. Some documents say that even with 1% packet loss can “significantly degrade” a VOIP call using G.711 or G.729 codec. But “1% or less” is the best answer for this question.
How does LLQ help ensure that voice quality is maintained in a converged network?
A. LLQ allocates minimum bandwidth guaranteed to voice traffic.
B. LLQ allocates a priority queue to voice traffic at a guaranteed rate.
C. LLQ allocates a priority queue and a minimum guaranteed bandwidth queue for voice.
D. LLQ ensures that all traffic is treated fairly and hence voice traffic is not severely impacted.
Low-latency queuing (LLQ) is used to give specific traffic classes higher priority when transmitting on the router’s WAN interface. Low Latency Queuing allows delay-sensitive data such as voice to be dequeued and sent first (before packets in other queues are dequeued), giving delay-sensitive data preferential treatment over other traffic.
The bandwidth given to an LLQ priority queue (PQ) is both the guaranteed minimum and policed maximum. This helps prevent the queue starvation that occurs with PQ.
In which two situations would a voice gateway be required? (Choose two)
A. To connect a corporate or branch location to an IP WAN
B. To connect a corporate or branch location using VoIP to the PSTN
C. To connect a Cisco Unified Communications Manager to a LAN
D. To connect a Cisco Unified Communications network to a PBX
E. To connect a corporate or branch location to a MAN
Answer: B D
What protocol is used to monitor and provide control information about the quality of an RTP session?
RTCP is used to monitor and provide control information about the quality of RTP streams but notice that RTCP only provides feedback on the quality of the transmission link. It does not make any guarantees concerning quality of service.
Which three are components of a dial plan? (Choose three)
A. Call legs
B. Endpoint addressing
C. centralized control
D. Call coverage
E. Digit manipulation
F. Decentralized control
Answer: B D E
We should understand what a dial plan is before talking about its components. In short, a dial plan is a collection of rules the call-processing agent uses to route calls. Below lists the components of a dial plan:
Endpoint addressing is the addressing scheme that is used to reach voice endpoints. For example, a company numbering plan might use four-digit extensions at each location and a three-digit site code. To call a phone at your own location, you would dial the four-digit extension. To call a phone at a remote company location, you would dial the site code and the extension.
Call coverage: Special groups of devices can be created to handle incoming calls for a certain service according to different rules, avoiding dropped calls. For example: top-down, circular hunt, longest idle, or broadcast groups are popular ones that you will see while learning CCNA Voice.
Digit manipulation: Digits can be manipulated prior to or after a routing decision has been made. In some cases, it is necessary to manipulate the dialed string before routing the call, for example, when you are rerouting over the PSTN a call originally dialed using the on-net access code, or when you are expanding an abbreviated code (such as 0 for the operator) to an extension.
Other two components of a dial peer are:
Calling privileges (or COR – class of service): Different groups of devices can be assigned to different classes of service, by granting or denying access to certain destinations or resources. For example, Employee phones might be allowed to reach only internal and local PSTN destinations, while Executive phones could have unrestricted PSTN access. The calling privileges assigned to a device are typically called class of service. In a Cisco voice
gateway, class of service is implemented by assigning Class of Restrictions (COR) to dial peers.
Path selection: Depending on the calling device, different paths can be selected to reach the same destination. Moreover, a secondary path can be used when the primary path is not available (for example, a call can be transparently rerouted over the PSTN during an IP WAN failure).
(Reference: CCVP – Implementing Cisco Voice Gateways and Gatekeeper & CVoice v6.0 Module 4 Lesson 1)
Refer to the exhibit.
|A||dial-peer voice 6000 voip|
|C||session protocol sipv2|
|D||session target ipv4:10.19.153.2|
The configuration shows a dial peer that points to Cisco Unity Express. Which line of configuration is incorrect?
We don’t have G.729ulaw, just G.711ulaw. G729 has 3 annexes that are G.729a, G.729b and G.729ab.
Refer to the exhibit.
How many discrete call legs are needed to set up a call between the POTS phone attached to router 1 and the phone in the PSTN?
We need four call legs as shown below
Refer to the exhibit.
Which inbound dial peer on CMERouter1 will be matched when phone extension 1234 places a call to 2010?
A. Voip dial peer 30
B. Default dial peer 30
C. None, which will cause the call to drop
D. Default dial peer
For CMERouter1 the “dial-peer voice 30 voip” will be matched for the outbound dial peer, not inbound one. When there is no dial-peer matched, the router will use the default dial peer.
Refer to the exhibit.
Which two types of dial peers are needed to complete this call end-to-end? (Choose two)
A. Serial dial peer
B. PSTN dial peer
C. POTS dial peer
D. Network dial peer
E. VoIP dial peer
Answer: C E
What is the relationship between a call leg and a dial peer?
A. A call leg is a virtual connection to set up a call whereas a dial peer is a physical connection to complete an end-to-end call.
B. The call leg and the dial peers are both logical connections used to complete an end-to-end call.
C. A call leg is a virtual connection that is set up and torn down before the dial peer is established.
D. The call leg and the dial peer are both physical connections used to complete an end-to-end call.
Which type of voice port will be most cost effective to allow the gateway to terminate two circuits from the PSTN or a PBX?
C. PRI T1
G. CAS T1
For PSTN and PBX connection, we need to use an analog interface type. E&M signaling is designed to connect directly to a PBX system that also supports E&M interfaces. Many PBX brands have E&M analog trunk cards that can operate as either the trunk circuit side or the signaling unit side and Cisco gateway does support E&M interfaces.
Which of the following is selected first for an incoming dial peer?
B. incoming called-number
D. pots port
First, the gateway attempts to match the called number with the incoming called-number. If no match is found, the router or gateway attempts to match the calling number of the call set-up request with the answer-address of each dial-peers. If no match is found, it attempts to match the calling number of the call set-up request to the destination-pattern of each dial-peer.
Notice that these steps are just applied for inbound dial peer.
Which protocol provides VoIP packet sequence numbering?
The RTP protocol provides end-to-end network functions and delivery services for delay-sensitive, real-time data, such as voice and video. It runs on top of UDP and provides these services:
- Payload-type identification
- Sequence numbering
- Time stamping
- Delivery monitoring
Identify the VoIP network component that provides CAC, bandwidth control and management, and address translation.
D. Call agent
A gatekeeper can perform these tasks:
Address translation: The gatekeeper translates alias addresses (e.g., E.164 telephone numbers) to Transport Addresses, using a translation table that is updated using Registration messages and other means.
Bandwidth control: The gatekeeper controls how much bandwidth a terminal may use. The gatekeeper provides the above functions for terminals and gateways that have registered with it.
Bandwidth management: Limits the number of concurrent accesses to IP internetwork resources (gatekeeper-based CAC for bandwidth management) (CAC: Call Admission Control).
Which three of the following are appropriate solutions to address latency issues in a VoIP network? (Choose 3)
A. Use dejitter buffers
B. Increase bandwidth
C. Fragment data packets
D. Prioritize voice packets
Answer: B C D
Notice that buffers give smoother audio playout but they does increase latency in VoIP network.
Which three headers are compressed by cRTP? (Choose 3)
A. Data link
Answer: B C D
Compressed Real-Time Transport Protocol (cRTP) compresses IP/UDP/RTP headers on low-speed serial links. We shouldn’t use cRTP on any high-speed interfaces as the price of CPU utilization is higher than the bandwidth savings.
Which of the following best describes a function of RTCP?
A. RTCP provides encryption, message authentication and integrity, and anti-replay service for voice streams
B. RTCP uses even-numbered UDP ports in the range 16,384-32,767 to transport voice payloads
C. RTCP provides out-of-band control information for an RTP flow
D. RTCP caches an RTP packet’s Layer 3 and Layer 4 headers in the routers at each end of a link, resulting in lower bandwidth demand for subsequent RTP packets
While the Real-time Transport Control Protocol (RTCP) sounds very inportant, its primary job is just statistics reporting, which includes
- Packet count
- Packet Delay
- Packet Loss
- Jitter (delay variations)
These types of information are useful but not as important as the actual RTP audio streams. Keep this in mind to configure RTCP & RTP streams correctly in the future.
Which two of the following VoIP gateway platforms are considered to be Integrated Services Routers (ISRs)? (Choose two)
A. Cisco 2600XM Series
B. Cisco 2800 Series
C. Cisco 3700 Series
D. Cisco 3800 Series
Answer: B D
We can hardly find a complete definition for the Integrated Services Routers but you can understand ISR as following:
“An ISR integrates other network features into the router other than just routing features. Used mostly in small offices on ADSL lines, they offer things like VPN, firewall, and encryption services.”
or another definition:
“First, the ISR routers are devices with a low-performance CPU when comparing them to the usual workstation/server processors from Intel or AMD.
Second, they are, as their name suggests it, “integrated services routers”, i.e. universal devices capable of performing many diverse networking functions, and that is true. However, even if a device can provide a particular service, it does not mean that it has unlimited power for providing it, and also if a device supports various features, it does not necessarily mean that you can have all of them turned on and expect that they all will perform well under a high load. The ISR routers are very flexible, however, they are still considered to be, at least from the throughput point of view, low-end routers. Their strength is the versatility, not the raw throughput.”
Some benefits of Integrated Services Routers:
1. The ISRs are more cost effective than their legacy equivalents, particularly when the network requirements map to an existing bundle.
2. The ISRs are faster (up to five times) and can handle quite a bit more memory than the legacy platforms. The base configurations also have more memory.
3. The ISRs are designed with the ability to run multiple concurrent services (FW, NAT, IDS, QoS, etc.) at wire-speed.
4. All the ISRs have TWO built-in LAN connections – FE or GE.
5. All the ISRs have an embedded HW VPN accelerator – It is always included, it is just a matter of buying a VPN enabled image to turn it on. If that is not fast enough, a VPN AIM can be added to further enhance VPN performance.
6. The HWIC enabled slots provide an impressive 400Mbps of dedi-cated bandwidth (the old WICs provided up to 8Mbps). This is great news for LAN uplinks and Ethernet Switch HWICs. The NME slots offer up to 1.2Gbps per module (the standard NM was only 600Mbps).
7. The EVM slot offers high density digital/analog voice ports.
8. All the ISRs with voice support have on-board DSP slots. There is no need to use a NM slot for a network module with DSPs for voice applications anymore – the on-board DSP slots can provide enough DSP resources for most common requirements.
9. All the ISRs with voice support can provide voice mail functionality with CUE (AIM and/or NM). CUE was not supported on the 1700 family.
10. All the ISRs that support voice can provide in-line power to Ethernet switch ports via a HWIC-ESW-POE or a NM-ESW-PWR (optional AC-IP power supply is required for in-line power).
11. Most ISRs provide some option for power supply redundancy. The 2811, 2821, 2851 and 3825 have a RPS connector and the 3845 can take a built-in redundant power supply.
12. For investment protection, the ISRs support most of the existing WICs, VICs, VWICs and NM modules (check the datasheets for de-tails).
13. SDM (Security Device Manager), included on all ISRs.