SkillTestPro | CCNP Collaboration 350-801 CLCOR Practice Test 7

CCNP Collaboration 350-801 CLCOR Practice Test 7 (Hard) - Troubleshooting SIP Elements - Call Setup and Teardown, SDP, DTMF, and Hold/Resume/Transfer

CCNP Collaboration Exam Questions

1 / 10

Which SIP message sequence represents the normal call setup process between two SIP endpoints after an INVITE is sent?

SUBSCRIBE followed immediately by a NOTIFY with no INVITE involved at all

INVITE followed immediately by a BYE with no other messages exchanged at all

OPTIONS followed immediately by a CANCEL with no call ever being attempted

PUBLISH followed immediately by a 500 Server Error with no call setup occurring

INVITE, followed by one or more provisional responses (such as 100 Trying and 180 Ringing), then a 200 OK final response, followed by an ACK from the calling party

2 / 10

Which SIP message is used by either party to gracefully terminate an already-established call?

A message used exclusively to update a user's voicemail greeting with no bearing on call state

A message used exclusively to request a directory listing with no bearing on call state

A message used exclusively to request a phone's firmware version with no bearing on call state

A message used exclusively to synchronize the phone's internal clock with no bearing on call state

BYE

A message used exclusively to report printer toner levels with no bearing on call state

3 / 10

Which protocol, carried within the body of SIP INVITE and response messages, negotiates the media capabilities (such as codecs, ports, and media type) for an audio or video call?

A protocol used exclusively for managing badge access control readers

A protocol used exclusively for transferring firmware update files between phones

A protocol used exclusively for tracking printer toner levels across the network

SDP (Session Description Protocol)

A protocol used exclusively for distributing voicemail greeting audio files

A protocol used exclusively for synchronizing phone directory photos

4 / 10

A call establishes audio successfully but video never connects between two SIP video endpoints, even though both devices support video calling. Which area should be investigated first?

The volume level configured on either endpoint's speaker

The SDP media line negotiation for the video stream (codec compatibility, port assignment, and whether the video m-line was rejected by either endpoint)

The color scheme configured on either endpoint's user interface

The wallpaper image configured on either endpoint's home screen

The ringtone assigned to either endpoint's primary line

The font used to display the caller's name on either endpoint

5 / 10

Which DTMF transport method, defined in RFC 2833/4733, carries digit tones as separate RTP payload events rather than embedding them directly in the voice audio stream?

RTP named telephone event (out-of-band DTMF relay)

A method that transmits DTMF exclusively via SMS text message

A method that transmits DTMF exclusively via a physical badge swipe

A method that transmits DTMF exclusively via email attachment

A method that transmits DTMF exclusively via a printed fax cover sheet

A method that transmits DTMF exclusively via postal mail

6 / 10

A user reports that pressing touch-tone digits during an IVR call is not being recognized by the far-end system, even though the call audio is otherwise clear in both directions. Which mismatch is a likely cause?

A mismatch in the ringtone assigned to the user's line

A mismatch in the wallpaper image configured on the user's phone

A mismatch in the color of the phone's keypad backlight

A mismatch in the volume level of the user's handset speaker

A DTMF transport method mismatch between the endpoints (for example, one side expecting RFC 2833/4733 out-of-band events while the other expects in-band or SIP INFO-based DTMF)

A mismatch in the font used to display the dialed digits on-screen

7 / 10

Which SIP mechanism is typically used to place an active call on hold, signaled by modifying the SDP media direction attribute?

A message that changes only the phone's ringtone with no SDP change

A re-INVITE (or UPDATE) that changes the SDP media direction attribute to sendonly or inactive for the held party

A message that changes only the phone's backlight timer with no SDP change

A BYE message sent to permanently terminate the call rather than hold it

A REGISTER message sent to re-register the endpoint with the call control server, in most deployment scenarios, which is not the standard approach

A message that changes only the phone's display wallpaper with no SDP change

8 / 10

Which SIP signaling flow describes a successful attended (consultative) transfer, at a high level?

The transferring party sends an OPTIONS request to the transfer target with no INVITE or REFER involved

The transferring party sends only a REGISTER message with no REFER or hold involved

The transferring party immediately sends a BYE to the original caller with no further signaling of any kind

The transferring party permanently merges both calls into a single indistinguishable audio stream with no separate legs

The transferring party places the original call on hold, establishes a new call to the transfer target, consults with them, and then uses a REFER (with Replaces) to connect the original caller directly to the transfer target while the transferring party's legs are torn down

The transferring party disables SIP signaling entirely and relies solely on manual conference room switching

9 / 10

Which troubleshooting step would be appropriate if a call resumes with no audio after being placed on hold and then resumed, even though the initial call had working two-way audio?

Examine the re-INVITE/UPDATE SDP exchange used for the resume operation to verify the media direction attribute correctly returned to sendrecv and that the negotiated codec and port information remain valid

Assume the issue is unrelated to SDP and only check the office's lighting

Assume the issue is caused by the phase of the moon

Assume the issue is caused by the phone's physical desk placement

Immediately replace the phone's hardware as the first troubleshooting step

Assume hold and resume is a deprecated capability no longer supported by SIP

10 / 10

Which statement accurately explains why SDP renegotiation is required during SIP hold, resume, and transfer operations rather than being handled purely at the SIP signaling layer?

SDP is only relevant during initial call setup and is never used again for the remainder of a call, under normal operating conditions, under typical configuration conditions, which is inconsistent with expected behavior, regardless of the platform version in use, regardless of the platform version in use

Hold, resume, and transfer operations never involve any change to how media is flowing

Because SDP describes the actual media stream parameters (direction, codec, ports), any change in how media should flow — such as pausing it for hold or redirecting it for transfer — must be reflected in a renegotiated SDP offer/answer exchange, not just in SIP call state alone

SDP and SIP signaling are simply two different names for the exact same identical protocol layer

SDP renegotiation is only required for video calls and never for audio-only calls

SIP signaling alone can redirect media streams without any SDP involvement in any scenario

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