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Cisco 300-815 Exam
Implementing Cisco Advanced Call Control and Mobility Services (CLASSM) (Page 8 )

Updated On: 12-Feb-2026
Page 8 of 50
PDF with 245 Questions

A customer reports audio quality issues between video endpoints in the HQ location in California and one of the branches in Texas.
Which two actions must RTCP take to troubleshoot this issue? (Choose two.)

  1. Allow for VAD to be used for calls using the G7.29 codec, which reduces the usage of the WAN bandwidth and saves around 30% of bandwidth per call
  2. Configure the rtcp keepalive command in Cisco Unified Border Element to generate reports, which can be reviewed using the debug voip rtcp packet command.
  3. Encrypt the media to stop rogue devices from replying and putting that traffic back on the WAN, which avoids any extra bandwidth and ensures the quality of the calls.
  4. Gather statistics on a media connection and information such as packets sent, lost packets, jitter, feedback, and round-trip delay. This information can help isolate the type of audio quality issues and the direction of the affected traffic.
  5. Compress the headers of RTP traffic to lower the bandwidth consumption over the WAN, which allows more calls with less bandwidth consumed.

Answer(s): B,D



A voice administrator enabled secure media between multiple Cisco UCM clusters that are connected via a SIP trunk.
Which configuration parameter in the SIP trunk must be enabled to allow encrypted media traffic to traverse the SIP trunk?

  1. Check the Allow iX Application Media check box in the SIP Profile.
  2. Check the SRTP Allowed check box in the SIP Trunk configuration.
  3. Check the RTP Allowed check box in the SIP Trunk configuration.
  4. Uncheck the RTP Allowed check box in the SIP Trunk configuration.

Answer(s): B

Explanation:

To allow encrypted media (Secure RTP or SRTP) to traverse a SIP trunk between Cisco UCM clusters, you must enable the SRTP Allowed option in the SIP Trunk configuration. This permits the trunk to negotiate and carry SRTP streams, ensuring media encryption between clusters.



Users report that calls to some locations do not work. The phone is calling, the recipient picks up, but it does not connect. These users are in the headquarters and registered to the company Cisco UCM with default values for calling (G.711). The problem locations are smaller remote locations with low bandwidth capacity, so the phones are configured with ILBC. Which command should be used to troubleshoot the issue?

  1. show dspfarm dsp active
  2. show dsp active
  3. debug voice dspfarm
  4. debug dsp all

Answer(s): C

Explanation:

When there is a mismatch in codec capabilities -- such as G.711 at headquarters and iLBC at remote sites -- a transcoder is needed. To troubleshoot codec negotiation and transcoding issues, the command debug voice dspfarm provides detailed real-time information about DSP resource usage and media processing, which is crucial in identifying why calls fail to connect despite being answered.



An engineer receives reports from users that calls to external participants on mobile phones are failing. The gateway is based on a server client setup.
When logs are analyzed, what is the last message expected before the media is established that Cisco UCM sends?

  1. established connection with an early SDP to the gateway, which must respond with 200 OK
  2. modified connection with SDP to the gateway, which must respond with 200 OK
  3. established connection with SDP to the gateway, which must respond with 200 OK
  4. forward connection with an early offer to the gateway, which must respond with 200 OK

Answer(s): C

Explanation:

In a server-client gateway setup, Cisco UCM sends an INVITE with SDP (an offer) to the gateway to establish the media path. The last signaling message from Cisco UCM before media is established is the INVITE with SDP, and the expected response from the gateway is 200 OK, confirming that the call setup is complete and media can flow.



Users of a Cisco UCME report that calls lose audio after they are forwarded to some users. Calling point-to- point seems to function fine, but depending on the type of phone, some calls lose audio when forwarded.
Looking at the trace header that shows Allow:

INVITE,BYE,CANCEL,ACK,INFO,PRACK,COMET,OPTIONS,SUBSCRIBE,NOTIFY,REFER,REGISTER

What is missing and is a reason that the SRTP stream stops working?

  1. EARLY-OFFER
  2. FORWARD
  3. UPDATE
  4. REDIRECT

Answer(s): C

Explanation:

The UPDATE method is used in SIP to modify session parameters (such as media) before a final response (like 200 OK) is sent. If the Allow header on a SIP INVITE does not include UPDATE, then SIP endpoints or gateways cannot renegotiate media parameters mid-dialog -- such as when a call is forwarded.

Without support for UPDATE, SRTP streams cannot be correctly re-established or modified during call forwarding, resulting in loss of audio.






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