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AT Conference provides global reservation-less conference calling, web conferencing, and operator-assisted conferencing. Other services include call recording, transcription service , application and desktop sharing , live polling, and presenter video-casting.
If the remote control software package supports audio transfer, the playback software can run on the remote computer, while the music can be heard from the local computer, as though the software were running locally. Co-Browsing: the navigation of the Web by several people accessing the same web pages at the same time. When session leader ...
While standard web conferencing software is designed to facilitate remote audio and video communication, it has too much latency for live musical performance. [ 1 ] [ 2 ] Connection-oriented Internet protocols subject audio signals to delays and other interference which presents a problem for keeping latency low enough for musicians to play ...
Google Meet uses proprietary protocols for audio and video stream control. Interoperability between Google Meet and SIP/H.323-based conferencing equipment and software is available for Google Workspace customers through third-party services. [59] Google Meet is optimized for low-bandwidth mobile networks through WebRTC and uses QUIC over UDP.
Vendor-hosted web conferencing is usually licensed as a service based on one of three pricing models: a fixed cost per user per minute, a monthly or annual flat fee allowing unlimited use with a fixed maximum capacity per session, or a sliding rate fee based on the number of allowed meeting hosts and per-session participants (number of "seats").
Enhanced Voice Services (EVS) is a superwideband speech audio coding standard that was developed for VoLTE and VoNR. It offers up to 20 kHz audio bandwidth and has high robustness to delay jitter and packet losses due to its channel aware coding [1] and improved packet loss concealment. [2] It has been developed in 3GPP and is described in 3GPP ...
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There are audio conferencing service providers that support wideband connections from these and other VoIP endpoints, while also permitting PSTN participants to join the conference in narrowband. sipXtapi is an open-source solution for VoIP media processing engine supporting wideband and HD voice that provides RTP and codecs through a plugin ...