VoIP Design Elements
Here you will find answers to CVoice – VoIP Design Element Questions
Question 1
Refer to the exhibit.
Lighthorse Equine Management would like to investigate converging voice and data on their existing T1 Frame Relay WAN link between New York and Atlanta. Currently the following list of applications are consuming no more bandwidth than what is listed on this segment of the network.
T1 link 1536 kbps
e-mail 75 kbps
internet 200 kbps
Oracle 500 kbps
FTP 250 kbps
Total 1025 kbps
The customer has allocated 25% of the WAN link for routing updates and other overhead. They would like to increase the number of samples encapsulated in each PDU to 40 ms. You have calculated 6 bytes of overhead for Frame Relay, no cRTP, and the use of the G.711 codec. How many simultaneous calls could be placed on this link?
A. 0 calls
B. 1 call
C. 2 calls
D. no more than 5 calls
E. no more than 10 calls
F. no more than 20 calls
Answer: B
Question 2
Refer to the exhibit. A QoS strategy has already been deployed on the LAN. Choose three WAN QoS best practices that should be used over the WAN link. (Choose three.)
A. Implement NBAR.
B. Implement admission control.
C. Mark voice traffic as EF in DSCP.
D. Mark voice traffic highest priority in 802.1p.
E. Use cRTP to maximize bandwidth utilization.
F. Configure access switches to trust traffic from IP phones.
Answer: B C E
Question 3
Refer to the exhibit. Users are not able to complete a call from 678-555-1212 to 770-555-1111. What is the correct diagnosis for the problem?
A. incorrect destination-pattern in router 1
B. incorrect POTS dial-peer statement in router 2
C. incorrect session-target statement in router 2
D. incorrect port statement in router 1 pots dial peer
E. missing no digit-strip on the voip dial peer in router 1
Answer: A
Question 4
A telemarketing firm needs to use number translation for incoming and outgoing calls. They have defined two translation profiles, one for incoming and one for outgoing calls. What can be used to simplify this task?
A. dial peer
B. voice port
C. hunt group
D. trunk group
E. source IP group
Answer: D
Question 5
Which command parameter specifies that the router should not attempt to initiate a trunk connection but should wait for an incoming call before establishing the trunk?
A. codec clear-channel
B. connection-trunk 404555…. answer-mode
C. voice-port 1/0:1
D. ds0-group timeslots 1-23 type ext-sig
Answer: B
Question 6
In a VoIP environment when speech samples are framed every 20 ms, a payload of 20 bytes is generated. Assuming a total packet length of 60 bytes, what is the length of the packet header if cRTP is deployed without redundancy checks?
A. 1 byte
B. 2 bytes
C. 3 bytes
D. 4 bytes
E. 20 bytes
F. 40 bytes
Answer: B
Question 7
You have set up a complex dial plan using translation rules. The following translation rule has been configured. What output would correspond to the test translation-rule command?
| translation-rule 1 rule 0 ^0.. 215550210 rule 1 ^1.. 215550211 rule 2 ^2.. 215550212 rule 3 ^3.. 215550213 rule 4 ^4.. 215550214 rule 5 ^5.. 215550215 rule 6 ^6.. 215550216 rule 7 ^7.. 215550217 rule 8 ^8.. 215550218 rule 9 ^9.. 215550210 |
A. test translation-rule 1 512
The replaced number: 21555021512
B. test translation-rule 1 555
The replaced number: 55521555021
C. test translation-rule 1 617
The replaced number: 61721555021
D. test translation-rule 1 910
The replaced number: 21555021910
Answer: A
Question 8
Which device is used to allow an H.323 stream to transit a firewall?
A. gatekeeper
B. gateway
C. proxy
D. MCU
Answer: C
Question 9
To hide its identity when initiating calls, Phone B requests that Server B place its calls for it. What kind of device is Server B?
A. proxy
B. redirect
C. registrar
D. user agent client
E. user agent server
Answer: A
Q7…Shouldn’t the correct answer be D
Nevermind I figured it out. it tacks the last 2 numbers to the end
Q1 and Q6. Can anybody explain me how it calculate, please.
ou yes, please explain Q1 & Q6. I have my exame at July 2th and I will get well prepared
For Q1, I just figured out like this way.
Step 1, try to figure out the bandwidth for 1 call.
G711 codec: (0.04*64000bps)/8=320 bytes
IP/UDP/RTP: 40 bytes
Frame Relay: 6 bytes
1000/40=25 packets/second
Total BW for overhead: 320+40+6=366 bytes
Total BW in 1 second: 366*25=9150 bytes = 73.2kbps
Step 2, try to figure out how many BW in WAN can be used for calls.
1536kbps *(1-25%) – 1025kbps = 127 kbps
So, just 1 call can be placed in this link.
Q6
Before RTP header compression the header has 40bytes. After compression, header has 2 or 4 bytes. It says, without redundancy checks. Then the header length is 2 bytes, if with redundancy, it is 4 bytes.
thx Oker
can u explain me Q7 please??
cvoice : Re question 7
if the ^ sign is present it means change any number beginning with whatever immediately follows the ^ sign. Then you have to append the remaining 2 digits in the dialled number to the end of the changed number.
A test translation-rule 1 512. Matches rule 5 ^5.. 215550215 then append the remaining 2 digits of the dialled number giving 21555021512 which is correct.
B test translation-rule 1 555. Matches rule 5 ^5.. 215550215 then append the remaining 2 digits of the dialled number giving 21555021555 which is wrong.
C test translation-rule 1 617. Matches rule 6 ^6.. 215550216 then append the remaining 2 digits of the dialled number giving 21555021617 which is wrong.
Now you have to be very careful not to fall into a trap with answer D. All the other rules state that if it begins with a “x” the last digit of the changed number becomes x but rule 9 says if it begins with a 9 the last digit in the cahnged number is a 0 prior to the final 2 digits being added.
D test translation-rule 1 910. Matches rule 9 ^9.. 215550210 then append the remaining 2 digits of the dialled number giving 21555021010 which is wrong. There should not be a 9 in this answer as the 9 is replace with a 0.
Answer A is the only correct answer given.
I hope this helps as its only my understanding of translation rules. Someone may care to correct me if they see it differently.
Q7. The rule states that the last 2 numbers of the changed number will always be the last 2 numbers of the dialled number. Only A and D match this criteria so you only have to qork out which of those 2 is correct and ignore the rest
Dunc – Thanx 4 ur explaination.
Will there be any Simulation questions in the CVOICE exam and if there are any than those Simulation question will be from which topic???
Can anyone please help me out as i am going to give my CVOICE exam on 20th September 2010 and will these questions that are given on voicetut will be enough to clear this exam????
i also have my exam on 20th sept..
any1 attempted this exam recently .. pls help
I am testing tomorrow as well. Good Luck!
For question 1 and 6, I found some good information to answer them:
VoIP – Per Call Bandwidth
The following protocol header assumptions are used for the calculations:
+ 40 bytes for IP (20 bytes) / User Datagram Protocol (UDP) (8 bytes) / Real-Time Transport Protocol (RTP) (12 bytes) headers.
+ Compressed Real-Time Protocol (cRTP) reduces the IP/UDP/RTP headers to 2or 4bytes (cRTP is not available over Ethernet).
+ 6 bytes for Multilink Point-to-Point Protocol (MP) or Frame Relay Forum (FRF).12 Layer 2 (L2) header.
+ 1 byte for the end-of-frame flag on MP and Frame Relay frames.
+ 18 bytes for Ethernet L2 headers, including 4 bytes of Frame Check Sequence (FCS) or Cyclic Redundancy Check (CRC)
Bandwidth Calculation Formulas
The following calculations are used:
+ Total packet size = (L2 header: MP or FRF.12 or Ethernet) + (IP/UDP/RTP header) + (voice payload size)
+ PPS = (codec bit rate) / (voice payload size)
+ Bandwidth = total packet size * PPS
Sample Calculation
For example, the required bandwidth for a G.729 call (8 Kbps codec bit rate) with cRTP, MP and the default 20 bytes of voice payload is:
+ Total packet size (bytes) = (MP header of 6 bytes) + ( compressed IP/UDP/RTP header of 2 bytes)
+ (voice payload of 20 bytes) = 28 bytes
+ Total packet size (bits) = (28 bytes) * 8 bits per byte = 224 bits
+ PPS = (8 Kbps codec bit rate) / (160 bits) = 50 pps
Note: 160 bits = 20 bytes (default voice payload) * 8 bits per byte
+ Bandwidth per call = voice packet size (224 bits) * 50 pps = 11.2 Kbps
Think of how retarded the average person is, and realize halve of them are stupider than that.
Sent from my iPhone 4G
B. connection-trunk 404555…. answer-mode
in which part of the track this command???
can someone give me an explanation about the question 7 please!!!!!!!!!!!!
My cousin recommended this blog and she was totally right keep up the fantastic work!
I dont understand why the 9 becomes a zero for D on Question 7.
So what exaclty is happening in these rules, are we adding the 9.. at the end of the number?? are we replacing the last 2 digit on the number????
A is the correct answer because this number 512^5..215550215 becomes 21555021512 all we added was the 12 at the end correct???
So on D 910^9.. 215550210 becomes 21555021910 but this should really turn into 010 and they are just adding the nine there to confuse you???
Im very confused about this can someone explain more preciselly? are we adding three digits at the end?? are we replacing digits at the end?
Thanks in advance
Ok never mind I understand now all we are doing is adding the last two digits at the end of the whole number indicated by the wild cards .. but not the actual explicitly defined number
910 ^9.. XXXXXXX translates to XXXXXXX10 the 9 gets dropped for some reason?
825 ^8.. XXXXXXX translates to XXXXXXX25 ???
Can someone confirm my guess??
NIce explanations!Thnx
yep, that is right.