You have enabled SIP Early Offers.
Which of the following is most likely the reason? (Select the best answer.)
A.
A service provider requires it.
B.
Analog audio must be compressed in the stream.
C.
Codec conversion is required.
D.
H.323 trunks are in use.
Explanation:
Of the available choices, you are most likely to enable Session Initiation Protocol (SIP) Early Offers if a public
switched telephone network (PSTN) service provider requires it. By default, Cisco Unified Communications
Manager (UCM) uses SIP Delayed Offers. UCM uses the SIP Offer/Answer model to establish SIP sessions.
SIP Offers, which are messages sent in the Session Description Protocol (SDP) fields of a SIP message,
contain information about the streams, codecs, IP addresses, and ports supported by the device. The remote
device receives the SIP Offer and sends a SIP Answer in the SDP fields of its response. SIP Early Offers are
sent by the session initiator in the SIP Invite, before the session is established. SIP Delayed Offers are sent by
the session initiator after the receiver sends its capabilities.
Delivering SIP Early Offers over SIP trunks is a function of Media Termination Points (MTPs). There are two
ways to enable Early Offers on SIP trunks in Cisco Unified Communications Manager (UCM): select the MTP
Required check box on the SIP trunk or select the Early Offer support for voice and video calls (insert MTP if
needed) check box on the SIP profile that is connected to the SIP trunk. If MTP is required, all outbound calls
will use MTP, as will calls operating on the same codec. If MTP is enabled as needed, it is inserted if the trunk
is incapable of sending complete information about its media capabilities in the SIP Invite message.
Converting digital audio from one codec to another is a function of a transcoder. Transcoders enable
communication between devices that support dissimilar audio codecs. For example, a transcoder can translate
a G.729encoded packet to a G.711encoded packet and vice versa. Transcoders translate the data stream in
real time between two devices so that no audio delay is experienced by either endpoint.
Sampling, encoding, and compression of analog audio is a function of a digital signal processor (DSP). DSPs
execute the steps required to convert an analog voice signal to digital packets, which allow voice data to
traverse a Voice over IP (VoIP) network. There are four steps involved in converting analog audio data to digital
audio data: sampling, quantization, encoding, and compression.
Providing connectivity for SIP devices and H.323 devices is a function of a UCM’s SIP trunks and H.323 trunks,
respectively, not of MTPs. UCM supports both SIP trunks and H.323 trunks.https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/9x/uc9x/trunks.html#wp1126439
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/7x/uc7_0/trunks.html#wp1045957