Protocols And Endpoints

You have a requirement to use secure signaling for SIP endpoints. Which protocol and port combination must be used in the SIP Security Profile?

You are troubleshooting intermittent audio quality. You notice that RTCP packets are arriving with high variance. What does this indicate?

When a SIP phone initiates a call, which message does the CUCM send to acknowledge the initial INVITE?

A network engineer is troubleshooting a Cisco IP Phone 7841 that fails to register with CUCM. The phone displays 'Configuring IP' and then 'Registering'. The engineer verifies that the phone has obtained an IP address and can ping the TFTP server. Which protocol does the phone use to download its configuration file from the TFTP server?

You are troubleshooting a scenario where G.711 is being used instead of the expected Opus codec between two endpoints in the same region. What is the most likely cause in the CUCM configuration?

You are troubleshooting a registration issue where the phone sends a REGISTER request, but the server does not respond. What is the most likely issue?

Which TWO features are provided by the Cisco Webex App when registered as a Softphone?

An endpoint sends a SIP REGISTER request. CUCM returns a 401 Unauthorized response. What is the next logical step in the SIP signaling flow for the endpoint?

A Cisco IP Phone 8845 in the New York office registers to CUCM and downloads its configuration. The administrator notices that the phone uses a DHCP scope option to discover the TFTP server address. Which DHCP option is used by the phone for this purpose?

When configuring an IP phone for automatic provisioning, which DHCP option is most commonly used to provide the TFTP server address to the phone?

A Cisco IP Phone 8845 registered to CUCM is not receiving its configuration file from the TFTP server after a factory reset. The phone has received its IP address via DHCP and can ping the TFTP server. Which option in the DHCP scope must be configured to allow the phone to locate the TFTP server?

A collaboration engineer is troubleshooting a one-way audio issue between two Cisco IP Phones registered to the same Cisco Unified Communications Manager cluster. The phones are in different locations separated by a WAN link. The engineer captures traffic and notices that the SIP signaling is correct, but RTP packets are only flowing in one direction. The engineer suspects a NAT or firewall issue. Which protocol or feature should the engineer verify to ensure that RTP can traverse the NAT and firewalls correctly?

Which protocol is primarily used for the transmission of real-time voice and video data between two endpoints?

A network administrator is configuring a Cisco IP Phone 7811 to register with Cisco Unified Communications Manager. The phone is powered on and connected to the network. The administrator notices that the phone is not obtaining an IP address. Which protocol is responsible for providing the phone with an IP address and other network parameters?

A collaboration engineer is deploying a new Cisco Webex Room Kit Pro in a conference room. The device is registered to Cisco Unified Communications Manager (CUCM) using SIP. The engineer needs to ensure that the device can place calls to external numbers and that the dialed digits are correctly processed. Which two configuration elements must be defined on CUCM to enable external dialing from this endpoint? (Choose two.)

Which protocol provides the signaling for Webex Room devices when they are registered to CUCM (on-premises)?

A Cisco Webex Room Kit is registered to CUCM on-premises. The administrator wants to enable encryption for signaling and media. Which protocol does the device use for secure signaling when registered to CUCM?

Which component in the Cisco Collaboration architecture provides the initial configuration files (XML) to an IP phone upon registration?

You are monitoring RTP traffic using a protocol analyzer. You observe that the Payload Type field in the RTP header is set to 0. Which codec is currently in use?

Which THREE network conditions contribute to poor VoIP call quality in an RTP stream?