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 the

Question

Options

• AGo to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 16384-32767.
• BAsk the firewall administrator to change the ports to TCP.
• CAsk the firewall administrator to change the range of UDP ports to 16384-32767.
• DGo to the SIP profile assigned to these IP phones in Cisco Unified CM and change the range of media ports to 20000-22000.
• EGo to System Parameters in Cisco Unified Communications Manager and change the range of media ports to 20000-22000.

• A
  5% (1)
• B
  11% (2)
• C
  79% (15)
• E
  5% (1)

Explanation

The root cause is a mismatch between the media (RTP) port range configured on the CUCM SIP Profile for the IP phones and the UDP port range permitted by the firewall. Cisco Unified CM's default RTP media port range in SIP Profiles is 16384–32767, which is much wider than the firewall's permitted range of 20000–22000. When RTP packets are sourced from ports outside 20000–22000, the firewall drops them, causing intermittent one-way or no audio. Two valid fixes are: (C) expand the firewall's UDP rule to cover 16384–32767 to match CUCM's default, or (D) narrow the SIP Profile's media port range to 20000–22000 to stay within the firewall's current rules. Option A is wrong because changing the SIP profile to 16384–32767 doesn't solve anything if the firewall only allows 20000–22000. Option E is wrong because media port ranges are set in SIP Profiles, not System Parameters.

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