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Plan for call admission control in Skype for Business ...
https://docs.microsoft.com/en-us/skypeforbusiness/plan-your-deployment/enterprise-voice-solution/call-admission-control
Peer-to-peer audio calls between Skype for Business endpoints will use either RTAudio (8kHz) or RTAudio (16kHz) when you factor in the bandwidth and prioritization of codecs. Conference calls between Skype for Business endpoints and the A/V Conferencing service will use either G.722 or Siren.
RTAudio - Wikipedia
https://en.wikipedia.org/wiki/RTAudio
The RTAudio encoder is capable of encoding single-channel (mono), 16 bit per sample audio signals. The encoder can be configured to operate either in the Narrow Band mode (8 kHz sampling rate) or the Wide Band mode (16 kHz sampling rate). The RTAudio decoder has a built-in jitter control module as well as an error concealment module. Licensing
A Practical Guide to Audio Codecs | No Jitter
https://www.nojitter.com/practical-guide-audio-codecs
RTAudio is an adaptive bandwidth codec that provides both wideband and narrowband audio. Additionally, it can adapt to network conditions and consume more or less bandwidth as needed. While it may not be a standard, RTAudio does a great …
Mediation Server and RTAudio - social.technet.microsoft.com
https://social.technet.microsoft.com/forums/lync/en-US/37deaf85-f398-4a24-97b2-57c479e4ba37/mediation-server-and-rtaudio
Hi, the RTaudio is dynamic so if Lync have enough bandwidth in a peer to peer call, lync will use normally RTAudio wideband. You can see the used codec in with the Monitoring Server on the Report for the user under media Quality Report, audio-stream, x-msrta is RTaudio and the sample-rate 16000 is 16khz for wideband.
Codecs Continued | Tao, Zen, and Tomorrow
https://andrewjprokop.wordpress.com/2013/09/25/codecs-continued/
The effective bandwidth of RTAudio wide band is 57 Kbps per call which is less than its G.722 counterpart. RTAudio can also work in narrow band consuming approximately 40 Kbps per call. Still More. This adds two more codecs to your arsenal of knowledge, but there are many more to go. However, if have a basic understanding of these five codecs (G.711, G.729, …
Video calls bandwidth calculation
https://social.microsoft.com/forums/en-US/1f41d808-6b79-4548-b68f-e5cee2ba4229/video-calls-bandwidth-calculation
OC 2007 client uses RTAudio Wideband encoded at an average bit rate of 29 kbps in 20 ms packets. By default OC 2007 calls are encrypted using SRTP, the per-packet overhead is approximately Ethernet II = 14 byte header + 4 byte footer, IP = 20 byte header, UDP = 8 byte header, RTP = 12 byte header, SRTP = 12 byte footer.
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