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In diesem Teil der IEC 61883 wird ein Protokoll für den Transport von unkomprimierten oder komprimierten Videodaten im Format 4:2:2 der Empfehlung ITU-R BT.601 (einschließlich kompatibler Erweiterungen auf dieses Format für höhere und niedrigere Auflösungen von anderen allgemein verwendeten Video-...
This Recommendation1 describes the methodology for deriving equipment impairment factors (Ies) using instrumental models. The equipment impairment factors derived by this methodology are intended to be used in the E-model (see [ITU-T G.107]). The methodology is to be considered as supplementary to...
This document specifies the procedures to test implementations of EVRC-A, EVRC-B, EVRC-WB, EVRC-NW or EVRC-NW2K compatible variable-rate speech codecs either by meeting the bit-exact implementation, or meeting recommended minimum performance requirements. The EVRC-A is the Service Option 3...
This document defines the Opus interactive speech and audio codec. Opus is designed to handle a wide range of interactive audio applications, including Voice over IP, videoconferencing, in-game chat, and even live, distributed music performances. It scales from low bitrate narrowband speech...
This document specifies Real-time Transport Protocol (RTP) payload formats to be used for the Enhanced Variable Rate Narrowband-Wideband Codec (EVRC-NW). Three media type registrations are included for EVRC-NW RTP payload formats. In addition, a file format is specified for transport of...
This Recommendation provides definitions, use cases and functional requirements for telepresence systems. A telepresence system is a set of functions, devices and network elements which are able to capture, deliver, manage and render multiple high quality interactive audio and video signals in a...
Service Options 3, 68, 70, 73 and 77 provide two-way voice communications between the base station and the mobile station using the dynamically variable data rate speech codec algorithm described in this standard. The transmitting speech codec takes voice samples and generates an...
Leading-edge VoIP technologies, tools, and standards. Efficiently deliver voice, data, and multimedia content over today's always-on broadband networks with guidance from this fully updated resource. Carrier-Grade VoIP, Third Edition, shows how to set up and administer a highly reliable, unified...
The RTP Profile for Audio and Video Conferences with Minimal Control (RTP/AVP) is the basis for many other profiles, such as the Secure Real-time Transport Protocol (RTP/SAVP), the Extended RTP Profile for Real-time Transport Control Protocol (RTCP)-Based Feedback (RTP/AVPF), and the Extended...
This Recommendation defines do-no-harm (DNH) tests for network-based and terminal-based voice quality enhancement (VQE) functions and non-VQE functions. These tests are applicable to functions used in both fixed and mobile networks and are independent of transport technology (e.g., TDM, ATM and IP)...
This document specifies a scheme for packetizing Standard apt-X or Enhanced apt-X encoded audio data into Real-time Transport Protocol (RTP) packets. The document describes a payload format that permits transmission of multiple related audio channels in a single RTP payload and a means of...
This specification defines an XML document format to describe the media properties of Session Initiation Protocol (SIP) sessions. Examples for media properties are the codecs or media types used in the session. This document also defines an XML document format to describe policies that limit...
Proxy servers play a central role as an intermediary in the Session Initiation Protocol (SIP) as they define and impact policies on call routing, rendezvous, and other call features. This document specifies a framework for SIP session policies that provides a standard mechanism by which a proxy can...
INTRODUCTION This technical report describes the software distribution supporting H.263 baseline (Profile 0) Level 45, and includes description of usage and sample config files. In order to characterize codecs in typical cdma2000®1 environments, [1] specifies simulation methodology (error...
Speex is an open-source voice codec suitable for use in VoIP (Voice over IP) type applications. This document describes the payload format for Speex-generated bit streams within an RTP packet. Also included here are the necessary details for the use of Speex with the Session Description...
This standard defines the VC-2 video compression system through the stream syntax, entropy coding, coefficient unpacking process and picture decoding process. The decoder operations are defined by means of a mixture of pseudo-code and mathematical operations. VC-2 is an intra frame video...
This document specifies how a stereoscopic 3D High Definition ("HDTV") video contribution system based on the MPEG-2 Transport Stream (TS) performs coding, multiplexing, and decoding. It defines constraints for the input image pair, the bitstream, the multiplexing, timing synchronization, use of a...
1.1 This specification covers commercial zinc alloys in ingot form for remelting for the manufacture of castings from the alloys as specified and designated as shown in Table 1. TABLE 1 Chemical and North American Color Code RequirementsA,B Composition, % UNS Z34510 Slush Casting Alloy A UNS Z30500...
Scope and object This International Standard specifies both Standard Definition and High Definition receivers for the DVB-T system. It concerns: • broadcasters, and • receiver manufacturers. The objective is to define: • how to provide broadcasts that are understood by all receivers and enable...
Several MIME type/subtype combinations exist that can contain different media formats. A receiving agent thus needs to examine the details of such media content to determine if the specific elements can be rendered given an available set of codecs. Especially when the end system has limited...
