The Digital Signal Processor (DSP) Resource Calculator is a web-based tool that helps calculate the number of DSPs required on the system. It also determines the number of calls that can be handled simultaneously, taking into consideration each of the Coder-Decoder (codec) standards. This tool requires information such as the platform type, Cisco IOS release number and Interface modules to calculate the DSPs. This document provides information on how to calculate DSPs for TDM Voice service and Video conferencing service.
Why do you need DSP?
No. of DSPs required = Voice Termination + Audio IP Services + Video IP services
1. Voice Termination
TDM Trunk Termination
H.320 TDM Video
2. Audio IP Services
Acoustic shock prevention
3. Video IP Services
DSP Media Resources
Audio Conferencing: Mixing RTP streams for multi-party conference bridges
Audio Transcoding and Transrating: Support multiple codecs on the same call (e.g., G.711-G.729A)
Video Conferencing: Mixing video streams for multi-party conferences
Cisco Unified Border Element: Media Enhancement, Acoustic Shock Prevention, Noise Reduction, Video Quality Metrics, Transcoding and Transrating
Voice Termination: Terminating TDM trunks and encoding, compressing and packetizing the voice.
Dear Team,We have a call center setup where the callers will be calling a support number and choose options (like 1 for sales, 2 for technical support, etc.). The callers will be connected to the agents. And the agent may do a blind transfer call to the i...
Hello, I know I have done this before, but am having trouble finding where and how to disable the "press any digit to be connected" when answering a Single Number Reach call. On CUCM 11.5.1SU6. Have a situation where it needs to just answ...
Hi All, We have a CMS cluster and one of the devices developed a h/w fault and was RMA'd. The replacement came with 2.1 s/w and was upgraded to 2.5.3 and a backup was restored. The device was sitting in our office during our COVID lockdow...
Hello I am configuring a SIP TRUNK from my CUCM to a Gateway through Internet and the call arrives to the Gatewayy and it's processed successfully. The problem here is that as I am using Internet to create my SIP trunk I am receiving a lot of unknown SIP ...