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Which of the following protocols will the IP phones use…

Refer to the diagram.

The IP phone at extension 1002 uses SCCP to communicate with the CME router. The IP phone at extension
1001 uses SIP to communicate with the CME router.
Which of the following protocols will the IP phones use to carry voice data during a call between extension 1001
and extension 1002? (Select the best answer.)

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A.
RTP

B.
SCCP

C.
SIP

D.
The phones will be unable to establish a call, because they do not use the same protocol to communicate
with CME.

E.
RTCP

Explanation:
In this scenario, the phones will use Realtime Transport Protocol (RTP) to carry voice data during a call
between extension 1001 and extension 1002. An IP phone can use Skinny Client Control Protocol (SCCP) or
Session Initiation Protocol (SIP) to communicate with Cisco Unified Communications Manager Express (CME).
The signaling conversation between a CME router and an IP phone controls many of the IP phone’s functions,
such as call initiation, call termination, and call waiting notification. CME can use SIP to communicate with oneIP phone and SCCP to communicate with another? however, once CME has completed its signaling
conversation with each IP phone, the phones will establish an RTP data stream between them to convey the
voice traffic, as illustrated in the following diagram:

Although the CME router is not directly involved in the voice data stream between the IP phones once the call
has been established, CME continues to communicate with the IP phones for other functions related to the
existing call and for any new calls. For example, if the CME router needs to notify extension 1001 of a second
incoming call, CME can initiate a new SCCP or SIP conversation with extension 1001 while the original call is
still active. This signaling conversation is in addition to the original SCCP or SIP conversation that CME used to
set up the call and the RTP conversation between the two IP phones engaged in the call.
Realtime Transport Control Protocol (RTCP) is not used to carry voice streams between devices. RTCP is
designed to report packet statistics between two devices engaged in an RTP session. Each RTP session is
monitored by a corresponding RTCP session. Each RTCP session is established on the oddnumbered User
Datagram Protocol (UDP) port following the evennumbered UDP port established by RTP. For example, if an
RTP session is established on UDP port 23456, a corresponding RTCP session is established on UDP port
23457.

https://www.cisco.com/c/en/us/support/docs/voice/voice-quality/5219-fix-1way-voice.html#basic
https://www.cisco.com/application/pdf/en/us/guest/tech/tk587/c1506/ccmigration_09186a008012dd36.pdf


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