Internet Telephony Service Provider
Here you will find answers to CVoice – Internet Telephony Service Provider Questions
Question 1
You work as a network technician , study the exhibit carefully. The Acme Corp. uses H.323 to place calls to their supplier RR Industries. Acme also has a voice connection to an ITSP for long distance over a SIP network. Which configuration should Acme use to deploy the CUBE?
A.
service voice voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to sip
allow-connections sip to h323
B.
service voice voip
allow-connections h323 to h323
allow-connections h323 to sip
C.
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
D.
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
Answer: C
Explanation
The Acme Corp connects to the ITSP via SIP Trunk and connects to RR industries via H.323. The Acme Corp itself uses H.323 so we have to enable protocol interworking with allow-connections commands:
allow-connections h323 to h323: allow Acme Corp to communicate with RR industries (in both ways)
allow-connections h323 to sip: allow Acme Corp to talk with ITSP (Acme Corp can talk and ITSP can hear but not vice versa)
allow-connections sip to h323: allow ITSP to talk with Acme Corp (Acme Corp can hear and ITSP can talk but not vice versa)
Notice that the configuration for H.323 and SIP interworking is unidirectional, thus if bidirectional interworking is required, you need to configure the mirror-matching statement as well.
Acme Corp doesn’t use SIP so we don’t need to configure “allow-connections sip to sip”.
Question 2
H.323 is an umbrella Recommendation from the ITU Telecommunication Standardization Sector (ITU-T) that defines the protocols to provide audio-visual communication sessions on any packet network. Which CUBE configuration will support H.323 protocol interworking and address hiding?
A.
voice services voip
h323 interworking
media flow-around
B.
voice services h323 to h323
h323 interworking
media flow-through
C.
voice services voip
allow-connections h323 to h323
media flow-around
D.
voice service voip
allow-connections h323 to h323
Answer: D
Explanation
Address hiding is a security feature of the CUBE which will hide the IP address of the originating gateway. This feature is turn on by default so we don’t need to set it.
A and B are not correct because the command “h323 interworking” doesn’t exist (moreover A uses “media flow-around” feature which will turn off the address hiding feature).
C is not correct because it uses “media flow-around” feature too.
Question 3
Refer to the exhibit. The Acme Corp. is deploying a CUBE. As a component of protocol interworking between RR Industries and the ITSP, they need to configure at least two dial peers. When the IP WAN is functional, Acme Corp. wants to use 5-digit dialing to RR Industries. Which three dial peers will complete the configuration for Acme Corp.? (Choose three)
A. dial-peer voice 50 voip
destination-pattern 50…
session protocol sipv2
session-target ipv4:192.168.100.100
B. dial-peer voice 1000 voip
destination-pattern 51…
session-target ipv4:192.168.100.100
C. dial-peer voice 91 voip
session protocol sipv2
destination-pattern 91T
session-target ipv4:10.1.100.1
dtmf-relay rtp-nte digit-drop h245-alphanumeric
D. dial-peer voice 91 voip
destination-pattern 91T
session-target ipv4:10.1.100.1
dtmf-relay rtp-nte digit-drop h245-alphanumeric
E. dial-peer voice 1000 voip
destination-pattern 51…
session-target ipv4:10.1.100.1
F. dial-peer voice 50 voip
destination-pattern 50…
session-target ipv4:172.16.14.6
Answer: B C F
Q3, there are two C’s and answers are B, C, F (no F).
Thanks for your detection. I updated it.
in Q1, can we use the command
allow-connection SIP to h323
Because in books and videos i have seen only,
allow-connection h323 to h323
allow-connection h323 to SIP
allow-connection SIP to SIP
but not the above command. Can u please explain?
Because the “allow-connection” command is unidirectional so we have to allow “SIP to h323″ & allow “h323 to SIP”. Surely you can use the “allow-connection SIP to h323″ command.
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dear voice tut
kindly tell me are these question still valid for 640-436 or not
Hello. And Bye.
Q3 – Dial-peer command in Option C is incorrect. It should be dial-peer voice 91 voip.
On Question three can someone explain why they are using 50.. as the destination? shouldn’t the extensions end 1000-1200 or 0100 0199 ??? this makes no sense to me.
Another question why do they decide to turn on sip v2 for the dial peer going out the trunk as opososed to the dial peer 1000 which is also sip but not enabled. Can someone clarify this please?
And lastly can someone digest the full meaning of dtmf-relay rtp-nte digit-drop h245-alphanumeric command and why that is necessary por favor?
Thanks in advance
Ok nm on the destination patterns, I asume they are striping the first two digits for some reason or another and thats when it made sense to me. 55 50… anyone kow why they strip the first two digits like that? or do we just have to asume that it happens automagically.
@CiscoCalisto: The question requires us: “Acme Corp. wants to use 5-digit dialing to RR Industries” but we notice that RR Industries use 7 digits (from 555-0100 to 555-0199). Therefore the solution here is to ignore the first “55″, just keeping the digits from 5-0100 to 5-0199 so we can use the destination-pattern of 50… After matching that dial-peer we can use a translation-rule to add the first “55″ before forwarding the packets.
Same for 51…, this is the dial-peer used to call from RR Industries to Acme Corp.
@R1: It’s a typing mistake. I updated it, thanks for your detection.
Ah yes now I understand fully, Mr. Tut thank you very much!
can some one tell me what is the reason the dial peer in choice “c” is destination-pattern 91T
Any reason to put 91 in front, T we can understand,
thanks to all for great effort and specially to voicetut.
@Imran, I think its because we generally use 9 as prefix to dial out to the pstn.
The “1″ is probably US country code.
Have you taken the exam yet? Any updates?
Hi Mr. tut, thank you for your support and help. In Q3 the option c & d are same except one command #session protocol sip v2. Could you please explain me the difference between the option c & d, if you can ?????
Thank you
@ronirock, Since the ITSP is using SIP, you would need to include the session protocol sipv2 dial-peer subcommand.
@cafe_blues
I have exam on this sunday, do some one have any suggestion. which dump should i use @ voice tut.
I feel so much happier now I understand all this. Thakns!