End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters. Calls to and from the PSTN work as expected. To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?
To troubleshoot users being unable to hear the remote party, the 'c=' line of the SDP content is the most relevant field. The 'c=' line contains connection information, including the IP address that will be used for the RTP (Real-time Transport Protocol) stream, which is crucial for audio transmission in SIP calls. If there's an issue with the IP address, it can prevent the audio stream from properly connecting, resulting in users being unable to hear each other.
Why would RTP traffic that is sent from the originating endpoint fail to be received on the far endpoint?
The RTP traffic would fail to be received on the far endpoint if the far-end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path. When the SDP data is altered, it can lead the RTP stream to be directed to an incorrect IP address or port, resulting in the failure of the receiving endpoint to get the RTP traffic.
An administrator is troubleshooting call failures on an H.323 gateway via the CLI. To see signaling for media and call setup, which debug must the Administrator turn on?
To troubleshoot call failures on an H.323 gateway and to see the signaling for media and call setup, the administrator should use the command 'debug H.225 asn1'. H.225 handles the control signaling for call establishment, call termination, and call control among H.323 entities in the network. The H.225 protocol includes H.225.0 call signaling and the RAS (Registration, Admission, and Status) protocol, necessary for seeing the required signaling information.
What is first preference condition matched in a SIP-enabled incoming dial peer?
In a SIP-enabled incoming dial peer, the first preference condition matched is 'incoming uri'. This is because the SIP protocol relies on URI (Uniform Resource Identifier) to identify the endpoints involved in the communication. The 'incoming uri' condition allows the router to match the SIP request based on the URI specified in the incoming SIP message, ensuring that the call routing is handled correctly based on the endpoint's identifier.
Cisco SIP IP telephony is implemented on two floors of your company. Afterward, users report intermittent voice issues in calls established between floors. All calls are established, and sometimes they work well, but sometimes there is one-way audio or no audio. You determine that there is a firewall between the floors, and the administrator reports that it is allowing SIP signaling and UDP ports from 20000 to 22000 bidirectionally. What are two possible solutions? (Choose two.)
The issue arises because the range of media ports configured on the Cisco Unified Communications Manager (CUCM) and the firewall do not match, leading to intermittent voice issues such as one-way audio or no audio. The correct solutions are to either change the range of UDP ports on the firewall to 16384-32767 to match the default media port range used by CUCM or adjust the media ports on the SIP profile of the IP phones within CUCM to the range allowed by the firewall, which is 20000-22000. This ensures that both the firewall and CUCM are configured to use the same port range for media, resolving the connectivity issues.