https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/cube- codec-basic.html

These are two parameters that influence the total number of supported conference participants on a Cisco IOS XE router that has DSP modules. A voice codec is a software algorithm that compresses and decompresses voice signals for transmission over a network. Different voice codecs have different bandwidth requirements and quality levels. A PVDM module is a hardware component that provides digital signal processing (DSP) resources for voice applications such as conferencing and transcoding. A PVDM module has a fixed session capacity, which is the maximum number of voice channels that it can support simultaneously.

The DTMF Events through SIP Signaling feature allows telephone event notifications to be sent through SIP NOTIFY messages, using the SIP SUBSCRIBE/NOTIFY method as defined in the Internet Engineering Task Force (IETF) draft, SIP-Specific Event Notification.

The provisioning modes/types are Off/Auto/CUCM/Edge/Spark/TMS/VCS.
https://www.cisco.com/c/dam/en/us/td/docs/telepresence/endpoint/ce95/dx70-dx80-administrator- guide-ce95.pdf#D1536209_DX70- DX80_Administrator_Guide_CE95.indd%3A.859768%3A307505

Cisco Collaboration System 12.x Solution Reference Network Designs (SRND):
The call processing service of Unified CM does support the deployment of a single cluster's call processing nodes across an IP WAN as long as the total end-to-end round-trip time between the nodes does not exceed 80 ms and an appropriate quantity of QoS-enabled bandwidth is provisioned. The maximum one-way delay between any two Unified CM servers should not exceed 40 ms, or 80 ms round-trip time.
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab12/collab12/models.html