Voice Design

Question 9 in this section reads as follows:

What are the two most likely driving forces motivating businesses to integrate voice and data into converged networks? (Choose two)

A. Voice has become the primary traffic on networks.
B. WAN costs can be reduced by migrating to converged networks.
C. Their PSTNs cannot deploy features quickly enough.
D. Data, voice, and video cannot converge on their current PSTN structures.
E. Voice networks cannot carry data unless the PRI circuits aggregate the BRI circuits.

The correct answers are listed as D and E.

I understand why D is correct but I’m having trouble understanding why ‘E’ is considered correct. I am using “Authorized Self-Study Guide: Designing for Cisco Internetwork Solutions (DESGN)” Second Ed. by Diane Teare.

On page 502 there is a section heading in bold titled “Drivers for Integrating Voice and Data Networks”. Under this section, I clearly see one of the correct answers from this question; ‘D’- Data, voice, and video cannot converge on their current PSTN structures. I don’t, however, see the second correct answer ‘E’- Voice networks cannot carry data unless the PRI circuits aggregate the BRI circuits.

Could someone please identify if ‘E’ is actually the other correct answer?

I actually think that answer ‘C’- “Their PSTNs cannot deploy features quickly enough” would be the more appropriate choice to this question in addition with answer ‘D’. This is due to the fact that it is listed along with answer ‘D’ as one of the Drivers for Integrating Voice and Data Networks in the text.

Someone please advise. Thanks!

I agree with you

From Cisco Press CCDA Official exam cert guide, 3rd ed

pg
ISDN Primary Rate Interface (PRI) is a digital trunk link used to connect to a phone switch.
A separate channel is used for common channel-signaling messages. (pg 503)

The ISDN BRI interface includes two 64-kbps B channels for voice or data and a separate 16-kbps
D channel that provides signaling for the interface. (pg 506)

Using these technical definitions choice E
“Voice networks cannot carry data unless the PRI circuits aggregate the BRI circuits.”
is technically correct.

Whether I think this is a likely driving force or not is irrelevant to the “official” Cisco answer on the test.

Question 6:

This is how I would do it:

Bandwidth per call = Total Packet Size * Packets Per Second

Bandwidth per call = (40+(2 or 4 because of cRTP)+8+20+6) * (16000(codec bit rate) / 320(voice payload size in bits))

Bandwidth per call = 78 bytes * 50

Bandwidth per call = 3900 bytes

Bandwidth per call = 31.2 kbps

Total calls allowed = 256 kbps / 31.2 kbps

Total calls allowed = 8.205

Therefore correct answer is 8!

@ teddiekgb
your calculation is total wrong.

follow my calculation

load is 40bytes, bit rate is 16kbbs. packet size will be 20ms.

that means each second, there will be 50 packets.

size of each packet is 40 + 6 that is 46 bytes. so each call will consume 2300 bytes

BW is 256kbbs, or 32000 bytes

32000/2300 = 13.9 that is why correct answer is 13. correct

BW / Call = ((L2 Header + IP/UPP/RTP Header) * Codec) / Voice Payload

Call = (BW * Voice Payload) / ((L2 Header + IP/UPP/RTP Header) * Codec)

Call = (256kbps * 40bytes) / ((6bytes + 2bytes + 40bytes) * 16)

Call = 13.333

Simpler still:

CCDA official cert guid: page 528. Theres a table showing you the features of codecs with various compressions etc. look at G.728 with CRTP. The bandwidth is recorded as 19 kbps.
256/19 is 13.47~.

u guys r retarded.

@teddiekgb

question 6’s answer is correct # of calls = 13. This is simple because with Codec G.728 @ 16kbps we can get 26 calls with cRTP compression. Hence with a BW of 256-kbps we get 13 calls…

the question 11…
how it was calculated?
if I try to do it, I get an completely different result:

768/X=(6 + 40) * 8 / 160
X= 768 * 160 / ((6 + 40) * 8 )
X= 333,9

What do I do wrong?

for question 11…

the calculation is very similar to question 6… for a connection with codec G.711, you get 6 calls out of a 512 kbps when cRTP (no compression) isn’t applied…

hence,

512kbps = 6 calls ==> 256kbps = 3 calls
768kbps = 512kbps + 256kbps = 6 calls + 3 calls = 9 calls in total

or 768kbps = 515kbps * 1.5 ==> # of calls = 6 calls * 1.5 = 9 calls

hi Duckie,

Thank you very much for your answer!
okay, your starting point is that:
512kbps = 6 calls
kindly tell me how you have calculated this?
if you have calculated it with following formel:
…..
Bandwidth = ((Layer 2 header) + (IP/UDP/RTP header)) * (Codec
bit rate) / (Voice payload size)
…..
then how did you do it?
or where you have found that?
Thank you very much!

from cisco CCDA self study guide Voice Design section you can find this table:

Codec Payload Size BW (kbps) BW with cRTP # of Calls (without/with cRTP)
on 512 kbps

G.711(64kbps) 160 83 68 6/7
G.726(32kbps) 60 57 36 8/14
….
….
G.728(16kbps) 40 35 19 14/26
G.729(8kbps) 20 18 8 28/64
….

please read the section mentioned above for more info.

sorry the format of the table is messed up after i submited the previous post

Codec —– Payload Size —– BW (kbps) —– BW with cRTP —– # of Calls (without/with cRTP)
————————————————————————————on 512 kbps

G.711(64kbps) –160 ———–83 ——————-68—————————— 6/7
G.726(32kbps) — 60———– 57——————- 36—————————– 8/14
….
….
G.728(16kbps)— 40 ————-35 ——————19—————————- 14/26
G.729(8kbps) —– 20———— 18 ——————8—————————— 28/64

Hi Duckie,

Thank you very much for your help!!

To melon:

No Calculator is allowed for the exam, you will be provided with a pen and erasable board to do your math/questions with

QUESTION 11
Given a VoIP network with these attributes:
Codec: G.711
WAN bandwidth: 768Kbps
Packet Header: 6 bytes
Payload: 160 bytes
CRTP: No
How many calls can be made?

A. 7 calls
B. 8 calls
C. 9 calls
D. 11 calls
E. 13 calls

I’ve been very confused about how to calculate the correct answer but after re reading the books i know its option C. Here is what you need to know:

For codec g.711 you use 64kbps
If you see “CRTP:No” it means that you are not using Compression for RTP which means that you are gonna use the default value which is 40 bytes when you calculate the total packet size with the following formula:

Total Packet Size: (layer 2 header) + (IP/UDP/RTP header) + (Voice Payload size)

Ok now we are gonna replace the formula with the values that we have:

Total Packet Size: 6bytes (for packet header) + 40 bytes (because we are not using CRTP, if you use CRTP is gonna be 2 bytes, check the desgn self study guide, second edition page 535 line 3) + 160bytes (payload)

Total packet size = 206 bytes or in bits is gonna be 206bytes * (8bits/1byte) = 1648bits = 1.648Kbps

Now we need to calculate the Voice packets per second (PPS)

PPS= (codec bit rate) / (voice payload size)
PPS= 64kbps / (160 bytes * (8bits/1byte))= 64kbps / 1280bits = 64kbps / 1.28kbps = 50

Now we calculate the Bw required per call:

Bw per call = Total packet size * PPS
Bw per call = 1.648 kbps * 50 = 82.4 kbps per call

Finally:

Calls = 768 kbps / 82.4 kbps per call = 9.32

So answer is option C, hopefully this is gonna help people that was really confused like i was =S, good luck!

FOR QUESTION 9 ….

FROM PAGE 374 of Designing for Cisco Internetwork Solutions (DESGN) Foundation Learning Guide

These events are driving convergence to Unified Communications networks:
■ Companies want to reduce WAN costs by migrating to integrated networks that can
carry any type of data efficiently.
■ Data has overtaken voice as the primary traffic on many voice networks.
■ The PSTN architecture that was built for voice is not flexible enough to carry data
well. The PSTN cannot create and deploy features quickly enough.
■ Data, voice, and video cannot be integrated on the current PSTN structure.

Erik I beelive that for G.711, the std. BW calculation was to take 64 kbps and tack on another 8 bkps for overhead, getting us to 72 kbps. For G.729, it’s the x3 rule (8 kbps x 3) for overhead (the overhead is more than the actual useful data because of increased collisions) giving us ~ 24 kbps of required BW. But I noticed you are talking 90 kbps for G.711 so my question is what are the other 18 kbps for? Just some extra margin for overhead (e.g. UDP IP headers)?Best

Oh my God! This is exactly what i am looking for. Your website is awsome. I am sending email to my friend suggesting her your website!

Q6

voice packet size = L2 Header + IP/UDP/RTP Header + payload = 6 + 2 (crtp compression) + 40 = 48 bytes = 384 bits
voice pacet per pecond = bit rate / voice payload = 16kbps/40*8b = 50 packet per second
voice bw = voice packet per second * voice packet size = 50 * 384 = 19.2k

calls = WAN bw / voice bw = 256k / 19.2k = 13.3 calls. So the answer is letter C.

#####################################################

Q11

voice packet size = L2 Header + IP/UDP/RTP Header + payload = 6 + 40 + 160 = 206 bytes = 1648 bits
voice pacet per pecond = bit rate / voice payload = 64kbps/160*8b = 50 packet per second
voice bw = voice packet per second * voice packet size = 50 * 1648 = 82.400k

calls = WAN bw / voice bw = 768k / 82.4k = 9.3 calls. So the answer is letter C.

Which term accurately describes a specific measure of delay often used to describe voice and video networks?

A Jitter
B Flucks
C Latency
D Reliability

Anyone? Anyone? Jitter or Latency? Help!

Jitter buddy….it only really applies to voice and video is the rationale.

Jitter is variability in delay

When motivating businesses, you are looking to decrease costs. So converging voice, video and data into one decreases costs. For question 9.

Thank you for some other fantastic post. Where else may anybody get that type of info in such a perfect way of writing? I have a presentation next week, and I’m on the search for such info.

Parse error: syntax error, unexpected T_IF in /home/nicetut/public_html/9tut.com/wp-includes/comment.php on line

Could you correct this (site for CCNA)

http://www.9tut.com

Thanks. Passed the CCDA exams with a ggod score. Quite some new questions did pop up..
Anyways the dumps on this site are valid.

to its B and D because, you do not know if all companys have the same BRI/PRI combo for voice/data… the books says one of the primary reasons if to same money and other is that current PSTN cannot carry data efficiently.

==============

Question 9

What are the two most likely driving forces motivating businesses to integrate voice and data into converged networks? (Choose two)

A. Voice has become the primary traffic on networks.
B. WAN costs can be reduced by migrating to converged networks.
C. Their PSTNs cannot deploy features quickly enough.
D. Data, voice, and video cannot converge on their current PSTN structures.
E. Voice networks cannot carry data unless the PRI circuits aggregate the BRI circuits.

Good Explanations.

About Q9, correct answer are B and D.

A Pri circuit (E1/T1) is structured to carry voice. E1 = 30B+D, if this is used for data you get 1984b/sec (timeslot 0 is not available).

If the Pri circuit (E1/T1) is left unstructured to carry data it is more efficient. E1 = 2048b/sec.

As E is the ‘driving force’ it is therefore the correct answer.
(answer B is true statement BECAUSE answer E is true)

Voice & Video can only converge with data before it reaches the PSTN, so VoIP is needed to converge in the private network – Answer D