Cisco 350-801 Exam - Topic 10 Question 72 Discussion

Actual exam question for Cisco's 350-801 exam

Question #: 72
Topic #: 10

Refer to the exhibit.

https://i.postimg.cc/C57TkczG/image.png

A call is falling to establish between two SIP Devices The called device answers with these SOP Which SOP parameter causes issue?

AThe calling device did not offer a ptime value

BThe media stream is set to send only

CThe payload for G.711ulaw must be 18.

DThe RTP port is set to 0.

Show Suggested Answer

Suggested Answer: D

The RTP port is used to send and receive media packets during a call. If the RTP port is set to 0, the called device will not be able to send or receive media packets, and the call will fail.

The other options are not correct because:

A) The calling device did not offer a ptime value: The ptime value is used to specify the amount of time between each media packet. If the calling device does not offer a ptime value, the called device will use the default value of 20 milliseconds.

B) The media stream is set to sendonly: The media stream is set to sendonly when the called device is only able to send media packets, and not receive them. This is not a problem, and the call will still succeed.

C) The payload for G.711ulaw must be 18: The payload for G.711ulaw is the type of media packet that is used. The payload must be set to 18 for G.711ulaw, but this is not a problem, and the call will still succeed.

by Kristeen at Mar 20, 2024, 09:35 AM

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Brittni

9 months ago

Media stream set to send only? Sounds like a misconfiguration.

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Charlena

9 months ago

G.711ulaw payload should be 18, but I think that's not the main issue.

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Noemi

10 months ago

Wait, does the ptime value really matter here?

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Sommer

10 months ago

Totally agree, that's a common mistake.

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Simona

10 months ago

The RTP port is set to 0. That's a big issue!

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Irene

10 months ago

I feel like the payload for G.711ulaw being set to 18 is a red herring. It seems like a technical detail that shouldn't stop the call from establishing.

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Andra

11 months ago

I practiced a similar question where the media stream was set to send only. I wonder if that could be the problem in this case too.

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James

11 months ago

I'm not entirely sure, but I remember something about ptime values being important for media negotiation. Could that be the issue here?

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Candra

11 months ago

I think the issue might be with the RTP port being set to 0. That seems like it would definitely cause a problem with the call setup.

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Janine

11 months ago

Ah, I think I've got it. The key is to identify the SDP parameter that is not compatible with the calling device's capabilities. Time to put my SIP knowledge to the test!

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Demetra

11 months ago

I'm a bit confused by the technical details here, but I'll try to break it down step-by-step and see if I can figure out the right answer.

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Fannie

11 months ago

Okay, let's see... the options mention the ptime value, media stream direction, payload type, and RTP port. I'll need to think through how each of those could impact the call setup.

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Iluminada

11 months ago

The question mentions the called device answering with certain SDP parameters, so I'll focus on those to see which one might be causing the issue.

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Marge

11 months ago

Hmm, this looks like a tricky SIP negotiation issue. I'll need to carefully analyze the SDP parameters to identify the potential problem.

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Carlene

11 months ago

I'm not too familiar with Catalog Client Scripts, so I'll have to eliminate the options I'm not sure about. I think option A is the safest bet here.

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Hollis

11 months ago

This question seems pretty straightforward. I think the answer is B - Sales force automation (SFA) since that's a tool specifically designed to manage the customer pipeline.

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Mitsue

2 years ago

I think both options A and D could potentially cause issues. We should consider them both.

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Stanton

2 years ago

But if the RTP port is set to 0, wouldn't that also prevent the call from establishing properly?

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Natalie

2 years ago

I disagree, I believe the problem lies with option D, the RTP port being set to 0.

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Stanton

2 years ago

I think the issue is caused by option A, the calling device not offering a ptime value.

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Vallie

2 years ago

I think the issue is with the payload for G.711ulaw needing to be 18. That could be causing the problem.

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Brandon

2 years ago

But if the media stream is set to send only, wouldn't that still allow the call to be established?

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Marquetta

2 years ago

I disagree, I believe the problem lies in the media stream being set to send only.

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Earleen

2 years ago

I agree with Brandon, having the RTP port set to 0 would definitely cause a problem.

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Brandon

2 years ago

I think the issue is with the RTP port being set to 0.

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Ricki

2 years ago

Haha, I bet the answer is 'D' - the RTP port set to 0. That's just begging for trouble!

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Lauran

2 years ago

You know, I'm not too familiar with SIP protocols, but the way I see it, the 'send only' media stream could be causing problems. That might be worth looking into.

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Leah

2 years ago

Okay, let's see. The called device is answering with some SDP parameters. I think the issue might be with the 'ptime' value, since the question asks about that specifically.

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Dalene

2 years ago

Hmm, this question seems a bit tricky. The exhibit shows some SIP parameters, but I'm not sure which one is causing the issue. Let me take a closer look.

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