/mcpCisco Meeting Server The Cisco Meeting Server conference bridge solution allows Ad Hoc, Meet-Me, Conference Now, and Rendezvous conferences. This conference bridge offers premises-based audio, video, and web conferencing, and works with third-party on-premises infrastructure. It scales for small or large deployments. You can add capacity incrementally as needed, to ensure that you can support the current and future needs of your organization. This conference bridge provides advanced interoperability. Any number of participants can create and join meetings from: • Cisco or third-party room or desktop video systems • Cisco Jabber Client • Cisco Meeting App (can be native or with a WebRTC compatible browser) • Skype for Business A minimum release of Cisco Meeting Server 2.0 is required to use the Cisco Meeting Server conference bridge. The Cisco Meeting Server supports SIP as the signaling call control protocol. It has a built in Web Server that allows for complete configuration, control, and monitoring of the system and conferences. The Cisco Meeting Server provides XML management API over HTTP. Cisco Meeting Server is enhanced to support AV1 codec and does not support H.265 video codec and Far End Camera Control (FECC). Note Video Encryption Unified Communications Manager supports encryption of audio, video, and other media streams so long as the individual endpoints involved in the communication also support encryption. Unified CM uses the Secure Real-Time Transport Protocol (SRTP) to encrypt the media streams. Some of the features include: • Support for SIP and H.323 endpoints • Support for encryption of main audio and video line while operating in Media Termination Point (MTP) passthru mode • Support for multiple encryption methods • Support for Session Description Protocol (SDP) crypto-suite session parameters in accordance with RFC 4568 To provide encrypted communications, encryption keys are exchanged between the endpoints and Unified Communications Manager during the SIP call setup. For this reason, the SIP signaling should be encrypted using TLS. During the initial call setup, the video endpoints exchange a list of encryption methods they support, select an encryption suite supported by both endpoints, and exchange encryption keys. If the endpoints cannot agree on a common encryption suite, the media streams are unencrypted and transported using the Real-Time Transport Protocol (RTP). Feature Configuration Guide for Cisco Unified Communications Manager, Release 15 and SUs 872 SIP Interoperability Cisco Meeting Server