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.)
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?
Answer(s): B
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?
Answer(s): C
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?
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,REGISTERWhat is missing and is a reason that the SRTP stream stops working?
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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