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TCP UDP Questions

September 2nd, 2015 in ROUTE 300-101 Go to comments

Question 1

Explanation

It is a general best practice to not mix TCP-based traffic with UDP-based traffic (especially Streaming-Video) within a single service-provider class because of the behaviors of these protocols during periods of congestion. Specifically, TCP transmitters throttle back flows when drops are detected. Although some UDP applications have application-level windowing, flow control, and retransmission capabilities, most UDP transmitters are completely oblivious to drops and, thus, never lower transmission rates because of dropping.
When TCP flows are combined with UDP flows within a single service-provider class and the class experiences congestion, TCP flows continually lower their transmission rates, potentially giving up their bandwidth to UDP flows that are oblivious to drops. This effect is called TCP starvation/UDP dominance.
TCP starvation/UDP dominance likely occurs if TCP-based applications is assigned to the same service-provider class as UDP-based applications and the class experiences sustained congestion.
Granted, it is not always possible to separate TCP-based flows from UDP-based flows, but it is beneficial to be aware of this behavior when making such application-mixing decisions within a single service-provider class.

Reference: http://www.cisco.com/c/en/us/td/docs/solutions/Enterprise/WAN_and_MAN/QoS_SRND/QoS-SRND-Book/VPNQoS.html

Question 2

Question 3

Explanation

TCP Selective Acknowledgement (SACK) prevents unnecessary retransmissions by specifying successfully received subsequent data. Let’s see an example of the advantages of TCP SACK.

TCP_ACK.jpgTCP (Normal) Acknowledgement TCP_SACK.jpg
TCP Selective Acknowledgement

For TCP (normal) acknowledgement, when a client requests data, server sends the first three segments (named of packets at Layer 4): Segment#1,#2,#3. But suppose Segment#2 was lost somewhere on the network while Segment#3 stills reached the client. Client checks Segment#3 and realizes Segment#2 was missing so it can only acknowledge that it received Segment#1 successfully. Client received Segment#1 and #3 so it creates two ACKs#1 to alert the server that it has not received any data beyond Segment#1. After receiving these ACKs, the server must resend Segment#2,#3 and wait for the ACKs of these segments.

For TCP Selective Acknowledgement, the process is the same until the Client realizes Segment#2 was missing. It also sends ACK#1 but adding SACK to indicate it has received Segment#3 successfully (so no need to retransmit this segment. Therefore the server only needs to resend Segment#2 only. But notice that after receiving Segment#2, the Client sends ACK#3 (not ACK#2) to say that it had all first three segments. Now the server will continue sending Segment #4,#5, …

The SACK option is not mandatory and it is used only if both parties support it.

The TCP Explicit Congestion Notification (ECN) feature allows an intermediate router to notify end hosts of impending network congestion. It also provides enhanced support for TCP sessions associated with applications, such as Telnet, web browsing, and transfer of audio and video data that are sensitive to delay or packet loss. The benefit of this feature is the reduction of delay and packet loss in data transmissions. Use the “ip tcp ecn” command in global configuration mode to enable TCP ECN.

The TCP time-stamp option provides improved TCP round-trip time measurements. Because the time stamps are always sent and echoed in both directions and the time-stamp value in the header is always changing, TCP header compression will not compress the outgoing packet. Use the “ip tcp timestamp” command to enable the TCP time-stamp option.

The TCP Keepalive Timer feature provides a mechanism to identify dead connections. When a TCP connection on a routing device is idle for too long, the device sends a TCP keepalive packet to the peer with only the Acknowledgment (ACK) flag turned on. If a response packet (a TCP ACK packet) is not received after the device sends a specific number of probes, the connection is considered dead and the device initiating the probes frees resources used by the TCP connection.

Reference: http://www.cisco.com/c/en/us/td/docs/ios-xml/ios/ipapp/configuration/xe-3s/asr1000/iap-xe-3s-asr1000-book/iap-tcp.html

Question 4

Explanation

Global synchronization occurs when multiple TCP hosts reduce their transmission rates in response to congestion. But when congestion is reduced, TCP hosts try to increase their transmission rates again simultaneously (known as slow-start algorithm), which causes another congestion. Global synchronization produces this graph:

TCP_Global_Synchronization.jpg

 

Global synchronization reduces optimal throughput of network applications and tail drop contributes to this phenomenon. When an interface on a router cannot transmit a packet immediately, the packet is queued. Packets are then taken out of the queue and eventually transmitted on the interface. But if the arrival rate of packets to the output interface exceeds the ability of the router to buffer and forward traffic, the queues increase to their maximum length and the interface becomes congested. Tail drop is the default queuing response to congestion. Tail drop simply means that “drop all the traffic that exceeds the queue limit. Tail drop treats all traffic equally and does not differentiate among classes of service.

Question 5

Explanation

When TCP is mixing with UDP under congestion, TCP flows will try to lower their transmission rate while UDP flows continue transmitting as usual. As a result of this, UDP flows will dominate the bandwidth of the link and this effect is called TCP-starvation/UDP-dominance. This can increase latency and lower the overall throughput.

Question 6

Question 7

Question 8

Explanation

If the speed of an interface is equal or less than 768 kbps (half of a T1 link), it is considered a low-speed interface. The half T1 only offers enough bandwidth to allow voice packets to enter and leave without delay issues. Therefore if the speed of the link is smaller than 768 kbps, it should not be configured with a queue.

Comments
  1. Dereje
    February 24th, 2015

    First Comment! I think the 300-101 exam start at Jan 29 so these exam from these days on wards.

  2. bangash
    March 11th, 2015

    i am going to appear for the CCNP Route. any one have passed with new patron.

  3. evdm
    April 7th, 2015

    Very good explanation digitaltut.

  4. Anonymous
    April 9th, 2015

    good luck folks.

  5. Anonymous
    April 14th, 2015

    @digitaltut, should not the answers to q5 be A, B, C? overall throughput should be the same.

  6. Anonymous
    May 3rd, 2015

    Jitter is always present, throughput is lowered for the TCP flows, because of UDP traffic

  7. help
    May 12th, 2015

    do i have to study all labs of 300-101 and 642-902 ?
    or only 300-101
    why this site is showing both exam codes ?
    should i study only 300-101 questions and sims ?
    please support

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    May 14th, 2015

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    June 26th, 2015

    You can use CBT nugget or Kevin Wallace Video Tutorials I liked it.

  12. Dereje
    June 26th, 2015

    Help Just understand that 642-902 no longer available on Cisco Exam pool. since the New 300-101 is Used from January 29, 2015 till now. So you have to read 300-101 exam objectives topic on any Cisco offical certs such as Authors Kevin Wallace i Used to read it. I found very helpful. with all DIKTA.
    thanks

  13. Nthgimp
    July 1st, 2015

    Hi all,
    I am planning do my CCNP route exam on coming Saturday any one please update me for exam purpose is it enough to work out the 301-101 questions from digitault?

  14. Sahil
    July 3rd, 2015

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  15. digitaltut
    July 24th, 2015

    @all: Because of copyrighted issue, we had to remove all the questions and answers. We can only keep the explanations. You can download all the questions and answers at: https://mega.co.nz/#!0pUxSZoJ!s293gEdQu4xLndoA7zZTq5ldia3gdZlrZqNxc_AgpFc
    or
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    July 27th, 2015

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  18. bellys
    July 28th, 2015

    Is it jsut me or are there no question on the page just explanaitions

  19. Spelunker
    August 9th, 2015

    I can also only see the explanations. The questions are not showing up.

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    September 3rd, 2015

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    September 4th, 2015

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  23. Ali
    September 16th, 2015

    Do we have questions for these explanation on this website?

  24. Ali
    September 16th, 2015

    Have any one tried these websites for download the questions?

    @all: Because of copyrighted issue, we had to remove all the questions and answers. We can only keep the explanations. You can download all the questions and answers at: https://mega.co.nz/#!0pUxSZoJ!s293gEdQu4xLndoA7zZTq5ldia3gdZlrZqNxc_AgpFc
    or
    http://www.4shared.com/office/-GBRNxjKce/ROUTE_July_2015.html

  25. Sabinay
    September 27th, 2015

    @ Digitaltut, the two links ” https://mega.co.nz/#!0pUxSZoJ!s293gEdQu4xLndoA7zZTq5ldia3gdZlrZqNxc_AgpFc
    or
    http://www.4shared.com/office/-GBRNxjKce/ROUTE_July_2015.html” doesn’t contain all question of Ip services, BGP and gre tunnel,can u update ur pdf please.

  26. digitaltut
    September 27th, 2015

    Please use this link to see newest PDF file: http://www.digitaltut.com/route-questions-and-answers

  27. Sabinay
    September 28th, 2015

    @ digitaltut, Thanks a lot….

  28. muaad
    October 4th, 2015

    there is no Q#8 in the questions pdf file .. any explanation. thank you

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    October 7th, 2015

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  30. pat
    January 17th, 2016

    Ok guys I came across this question but things don’t add up, at least according to my understanding, see the below

    If the total bandwidth is 64 kbps and the RTT is 3 seconds, what is the bandwidth delay
    product?
    A. 8,000 bytes
    B. 16,000 bytes
    C. 24,000 bytes
    D. 32,000 bytes
    E. 62,000 bytes
    Answer: C – this is apparently the right answer

    but then Bandwidth Delay Product = bandwidth (bits per sec) * round trip time (in seconds)

    can anybody explain what I’m missing here

  31. vic
    January 19th, 2016

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  35. stubby
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    @route: vic’s link opened as a blog site with helpful links on the CCNP labs

  36. azim
    February 1st, 2016

    Bandwidth Delay Product = bandwidth (bits per sec) * round trip time (in seconds)

    Lets calculate: (64*1000) * 3 bit =192,000 bits = 192,000/8 bytes=24,000 bytes

  37. Anonymous
    February 18th, 2016

    thanks Azim much appreciated

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  40. Anonymous
    April 2nd, 2016

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  41. Druid
    April 24th, 2016

    Hello, question 8…

    Fragmentation has to do with MTU and not with queing. Would D be the correct answer? In a slow-link, putting the voice packets for queuing will increase latency

  42. TALAW
    May 16th, 2016

    Guys please help out ccnp route study material preferably pdf please which site can I get help

  43. to PAT
    May 26th, 2016

    BDP (bits) = total_available_bandwidth (bits/sec) x round_trip_time (sec)

    BDP= 64000*3=192000 bits=192000/8 bytes= 24000 bytes

  44. Anonymous
    July 19th, 2016

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  50. about Q8
    September 14th, 2016

    Why is the answer A ?
    When video traffic flow to the queue on a narrow link ,
    the traffic typically is not fragmented .
    what does it mean ?

  51. draugir
    September 16th, 2016

    guys, can anyone confirm that this questions are from CCNP R&S. I think this questions are from voice exam…

  52. Disappointed
    October 4th, 2016

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  53. im
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