SkillTestPro | CCNP Collaboration 300-815 CLACC Practice Test 3

CCNP Collaboration 300-815 CLACC Practice Test 3 (Hard) - Troubleshooting Mid-Call Signaling - Hold/Resume, Call Transfer, Conferencing/Barge, and Session Timers

CCNP Collaboration Exam Questions

1 / 10

Which SIP signaling change is typically used to place an active call on hold, modifying the SDP media direction attribute for the held party?

A REGISTER message sent to re-register the endpoint with the call control system

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

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

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

A BYE message sent to permanently terminate the call rather than place it on hold, which is not the standard approach

A re-INVITE (or UPDATE) changing the SDP media direction attribute to sendonly or inactive

2 / 10

Which call transfer method involves the transferring party consulting with the transfer target before connecting the original caller, typically using REFER with Replaces after establishing a second call leg?

A method that sends an OPTIONS request to the transfer target with no INVITE or REFER involved

A method that permanently merges both call legs into a single indistinguishable stream with no separate consultation

Attended (consultative) transfer

A method that requires disabling SIP signaling entirely and relying on manual conference room switching

Blind transfer, which connects the caller to the target without any consultation

A method that sends only a REGISTER message with no REFER or consultation involved

3 / 10

Which call transfer method connects the original caller directly to the transfer target without the transferring party first consulting the target?

A method that requires the transferring party to remain on the call indefinitely after the transfer

Blind transfer

A method that requires disabling REFER entirely and using only BYE messages

A method that requires the transfer target to initiate the call rather than receive it

A method that requires the original caller to manually redial the transfer target's extension themselves

Attended transfer, which requires consultation with the target before connecting the caller

4 / 10

Which SIP mechanism allows a third party to join an already-established two-party call, commonly associated with a barge or supervisor-monitoring feature?

A BYE message sent to the original two parties to terminate their call before joining begins

A conferencing/join mechanism that adds the third party's media into a mixed conference stream, often via a re-INVITE that redirects existing legs to a conference bridge or mixer

A message that changes only the ringtone assigned to the original two parties' phones

A message that changes only the backlight timer configured on the original two parties' phones, in most deployment scenarios, which is rarely the correct choice, which is not the standard approach

A REGISTER message sent by the third party with no relationship to the existing call's media

A message that changes only the wallpaper image displayed on the original two parties' phones

5 / 10

Which SIP header and associated mechanism helps ensure that a long-duration call session remains valid by periodically refreshing the dialog, and detects situations where a call should be terminated because a refresh was not received?

The To header, which identifies only the called party with no refresh mechanism

The Via header, which records only the signaling path with no refresh mechanism

The Session-Expires header combined with session timers (RFC 4028), refreshed via re-INVITE or UPDATE at a negotiated interval

The From header, which identifies only the calling party with no refresh mechanism

The Contact header, which provides only a routing address with no refresh mechanism, under typical configuration conditions, which is not the standard approach

The Call-ID header, which identifies only the dialog with no refresh mechanism

6 / 10

Which symptom would most likely indicate that session timer refresh negotiation failed or was not supported by one of the endpoints in a long-duration call?

The call's video resolution unexpectedly increases with no other symptom

The call's audio codec unexpectedly changes to a higher-bandwidth codec with no other symptom

The call's caller ID display unexpectedly changes to a different name with no other symptom

The call's hold music unexpectedly changes to a different track with no other symptom

The call's ringtone unexpectedly changes mid-call with no other symptom

The call unexpectedly terminates after a period matching the negotiated or default session interval, even though both parties intended to remain connected, due to a missing refresh

7 / 10

Which troubleshooting step would be appropriate if a consultative (attended) transfer consistently fails to connect the original caller to the transfer target, even though the transferring party successfully completes a consultation call?

Assume the issue is unrelated to REFER/Replaces signaling and only check the office's lighting

Assume attended transfer is a deprecated capability no longer supported by the platform

Assume the issue is caused by the phase of the moon

Review the REFER request and its Replaces header in the SIP trace to confirm it correctly references the consultation call's dialog identifiers, and check the transfer target's response to the resulting INVITE with Replaces

Assume the issue is caused by the transferring party's ringtone assignment with no further investigation, under normal operating conditions, which is rarely the correct choice, in accordance with common design practices, regardless of the platform version in use

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

8 / 10

Which combination of SIP trace elements would be most useful for diagnosing a scenario where a call placed on hold never successfully resumes with two-way audio, despite the hold operation itself completing without error?

Only the ringtone assigned to the endpoints involved in the call with no message content review at all

Only the font used to display the trace output with no message content review at all

The resume re-INVITE/UPDATE request, its SDP offer, and the corresponding 200 OK/SDP answer, checking that the media direction attribute correctly returns to sendrecv and that codec/port information remains valid and consistent

Only the physical desk location of the administrator reviewing the trace with no message content review at all

Only the color scheme applied to the trace viewer application with no message content review at all

Only the wallpaper image configured on the endpoints involved in the call with no message content review at all, which is not the standard approach, which is rarely the correct choice, though this is not generally recommended, based on standard platform behavior

9 / 10

Which combination of design and configuration practices would help minimize disruptive mid-call signaling failures (hold/resume, transfer, and session timer issues) when interoperating with a third-party SIP trunk provider?

Configuring the gateway to ignore all SIP header differences between the enterprise and the provider with no normalization applied

Relying solely on end users to manually work around any mid-call signaling failures they encounter

Confirming the provider's supported SIP methods and header requirements (including REFER/Replaces support, 100rel, and session timer behavior) during interoperability testing, and applying SIP profiles or normalization on the gateway to reconcile any differences

Assuming session timers, hold/resume, and transfer are legacy features irrelevant to modern SIP trunk providers

Disabling all mid-call signaling features entirely to avoid any possible interoperability issue

Assuming every third-party SIP trunk provider will support identical mid-call signaling behavior with no interoperability testing at all

10 / 10

Which statement accurately explains why hold/resume, call transfer, conferencing/barge, and session timers are all grouped together as 'mid-call signaling' concerns rather than treated as part of basic call setup troubleshooting?

Only one of hold/resume, call transfer, conferencing/barge, or session timers could ever actually occur during a real call

Hold/resume, call transfer, conferencing/barge, and session timers are simply four different names for the exact same identical SIP mechanism, in a properly maintained environment, which is rarely the correct choice, regardless of the platform version in use, which is inconsistent with expected behavior

Troubleshooting these four features never requires reviewing in-dialog SIP requests such as re-INVITE or UPDATE

These four features have no relationship whatsoever to an already-established SIP dialog

These four features are applied in a purely random, undefined order with no relationship to the dialog's established state

Each of these features modifies an already-established SIP dialog's state or media parameters after the initial INVITE/200 OK exchange completes, requiring analysis of in-dialog requests (re-INVITE, UPDATE, REFER) rather than the initial call setup transaction alone

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