Media traffic bandwidth - cp-mlxprod …



Session Initiation Protocol–Based FoundationSkype for Business Server 2015 is built on SIP-based architecture similar to most modern enterprise Voice over Internet Protocol (VoIP) and Unified Communications solutions. Although SIP has gained industry-wide acceptance and is more than ten years old, it is still a fairly new standard. Various Internet Engineering Task Force (IETF) working groups are working to further standardize and develop complementary solutions. SIP-related request for comments (RFCs) and the IETF working group documents are the foundational design elements underlying the Microsoft Unified Communications solution. To facilitate interoperability and interfacing with other systems, Skype for Business Server 2015 has been built on standards, wherever possible. SIP Definition The abstract of RFC 3261 defines SIP as an application-layer control or signaling protocol for creating, modifying, and terminating sessions between one or more participants. These sessions include Internet-based telephone calls, multimedia distribution, and multimedia conferences. You can use SIP to set up media sessions of any kind and not just telephony, modify the sessions while they are on, and then terminate the connections after the sessions are complete. For example, a participant can start an instant messaging session with another participant, add audio, and then video to the existing call, and finally terminate it. There is more to SIP than just handling media; SIP can be extended to perform multiple tasks. For example, SIP can manage the publishing and requesting of Presence information and the delivery of instant messages.Because of the dynamic evolution of Unified Communications, Skype for Business Server 2015 and similar products are not just based on RFC 3261, but also on the 200 or more IETF Internet drafts and proposed standards, and SIP-related RFCs. Unified Communications products are based on a range of these RFCs. The following table describes some drafts and standards that Skype for Business Server 2015 is built on.RFC/DraftDescriptionRFC 2782This is a DNS resource record (DNS RR) for specifying the location of services, for example, DNS service (SRV) records, and is used to locate servers and services in Skype for Business Server 2015.RFC 3428This is about the SIP for instant messaging and Presence leveraging extension (SIMPLE) that is used for IM conferencing.RFC 3966This is about the URI for telephone numbers. It defines how phone numbers should be represented in SIP communications.RFC 5239This is a framework for centralized conferencing (XCON) and is the architecture behind audio/video conferencing and web conferencing in Skype for Business Server 2015.ICE draft vg& draft v19This is about ICE, a protocol for Network Address Translation (NAT) traversal for offer/answer. These Internet drafts for ICE are used in Microsoft? Office Communications Server 2007, Microsoft? Communications Server 2007 R2, and Skype for Business Server 2015.There are many more RFCs and if you are interested in knowing more about the standard and non-standard protocols that are used, you can read the Microsoft Office protocol documents on the Microsoft website.Media traffic bandwidthThe media traffic bandwidth usage can be challenging to calculate because of the number of different variables, such as codec usage, resolution, and activity levels. The bandwidth usage is a function of the codec that is used and the activity of the stream, which can vary between scenarios. The following table lists the audio codecs typically used in Skype for Business Server scenarios.Audio codec bandwidth:Audio codec Scenario Audio payload bit rate (KBPS) Bandwidth audio payload and IP header only (Kbps) Bandwidth audio payload, IP header, UDP, RTP and SRTP (Kbps) Bandwidth audio payload, IP header, UDP, RTP, SRTP and forward error correction (Kbps) RTAudio WidebandPeer-to-peer29.045.057.086.0RTAudio NarrowbandPeer-to-peer PSTN11.827.839.851.6G.722Conferencing64.080.095.6159.6G.722 StereoPeer-to-peer Conferencing128.0144.0159.6223.6G.711PSTN64.080.092.0156.0SirenConferencing16.032.047.663.6SILK Wideband Peer-to-peer36.052.064.0100.0SILK Wideband Peer-to-peer26.042.054.080.0SILK Wideband Peer-to-peer20.036.048.068.0SILK wideband/narrowbandPeer-to-peer13.029.041.054.0The bandwidth numbers in the previous table are based on 20ms packetization (50 packets per second) and for the Siren and G.722 codecs include the additional secure real-time transport protocol (SRTP) overhead from conferencing scenarios and assume the stream is 100% active. Forward Error Correction (FEC) is used dynamically when there is packet loss on the link to help maintain the quality of the audio stream. The stereo version of the G.722 codec is used by systems that are based on the Lync Room System, which uses a single stereo microphone or a pair of mono microphones to allow listeners to better distinguish multiple speakers in the meeting room.Video Resolution Bandwidth:Video codec Resolution and aspect ratio Maximum video payload bit rate (Kbps) Minimum video payload bit rate (Kbps) H.264320x180 (16:9)212x160 (4:3)25015H.264/RTVideo424x240 (16:9))320x240 (4:3350100H.264480x270 (16:9)424x320 (4:3)450200H.264/RTVideo640x360 (16:9)640x480 (4:3)800300H.264848x480 (16:9)1500400H.264960x540 (16:9)2000500H.264/RTVideo1280x720 (16:9)2500700H.2641920x1080 (16:9)40001500H.264/RTVideo960x144 (20:3)50015H.2641280x192 (20:3)1000250H.2641920x288 (20:3)2000500The default codec for video is the H.264/MPEG-4 Part 10 Advanced Video Coding standard, together with its scalable video coding extensions for temporal scalability. To maintain interoperability with legacy clients, the RTVideo codec is still used for peer-to-peer calls between Skype for Business Server and legacy clients. In conference sessions with both Skype for Business Server and legacy clients the Skype for Business Server endpoint may encode the video using both video codecs and send the H.264 bitstream to the Skype for Business Server clients and the RTVideo bitstream to legacy clients.The bandwidth required depends on the resolution, quality, frame rate, and the amount of motion or change in the picture. For each resolution, there are two pertinent bit rates:Maximum payload bit rate???This is the bit rate that an endpoint will use for resolution at the maximum frame rate. This is the value that allows the highest video and sound quality.Minimum payload bit rate???This is the bit rate below which a Skype for Business Server endpoint will switch to the next lower resolution. To guarantee a certain resolution, the available video payload bit rate must not fall below this minimum bit rate for that resolution. This value is helps you understand the lowest value possible if the maximum bit rate is not available or practical. For some users, such a low bit rate video might provide an unacceptable video experience so use caution with these minimum video payload bitrates. Note that for static, unchanging video scenes the actual bit rate may temporarily fall below the minimum bit rate.Skype for Business Server supports many resolutions. This allows Skype for Business Server to adjust to different network bandwidth and receiving client capabilities. The default aspect ratio for Skype for Business Server is 16:9. The legacy 4:3 aspect ratio is still supported for webcams which don’t allow capture in the 16:9 aspect ratio.Video FEC is always included in the video payload bit rate when it is used so there are no separate values for with video FEC and without video FEC. Endpoints do not stream audio or video packets continuously. Depending on the scenario there are different levels of stream activity which indicate how often packets are sent for a stream. The activity of a stream depends on the media and the scenario, and does not depend on the codec being used. In a peer-to-peer scenario: Endpoints only send audio streams when the users speak.Both participants receive audio streams.If video is used, both endpoints send and receive video streams during the call.For static video scenes the actual bit rate may temporarily be very low as the video codec will skip encoding regions of the video without a change since the prior sample.In a conferencing scenario:Endpoints send audio streams only when the users speak.All participants receive audio streams.If video is used, all participants can receive up to five receive video streams and one panoramic (for example, aspect ratio 20:3) video stream. By default the five receive video streams are based on active speaker history but users can also manually select the participants from which they want to receive a video stream. If multi-video is enabled, the resolution and bandwidth requirement for each of the video streams will be lower.Each participant that turns on the user’s send video stream will send one or more video streams. Skype for Business Server has the capability of sending up to five video streams to optimize the video quality for all the receiving clients. The actual number of video streams being sent is determined by the sender based on CPU capability, available uplink bandwidth, and the number of receiving clients requesting a certain video stream. The most common case is that one H.264 and one RTVideo video stream are being sent in case a legacy client joins the conference. Another common scenario is that several H.264 video streams (for example, with different video resolutions) are sent to accommodate different receiver requests. In addition to the bandwidth required for the real-time transport protocol (RTP) traffic for audio and video media, bandwidth is required for real-time transport control protocol (RTCP). RTCP is used for reporting statistics and out-of-band control of the RTP stream. For planning, use the bandwidth numbers in the following table for RTCP traffic. These values represent the maximum bandwidth used for RTCP and are different for audio and video streams because of differences in the control data RTCP BandwidthMedia RTCP maximum bandwidth (Kbps) Audio5Video (Only H.264 or RTVideo being sent/received)10Video (H.264 and RTVideo being sent/received)15For capacity planning, the following two statistics are of interest:Maximum bandwidth without FEC???The maximum bandwidth that a stream will consume. This includes the typical activity of the stream and the typical codec that is used in the scenario without FEC.?This is the bandwidth when the stream is at 100% activity and there is no packet loss triggering the use of FEC.? This is useful for computing how much bandwidth must be allocated to allow the codec to be used in a given scenario.? FEC is not expected to be a requirement on a managed network.Maximum bandwidth with FEC???The maximum bandwidth that a stream consumes. This includes the typical activity of the stream and the typical codec that is used in the scenario with FEC. This is the bandwidth when the stream is at 100% activity and there is packet loss triggering the use of FEC to improve quality. This is useful for computing how much bandwidth must be allocated to allow the codec to be used in a given scenario and allow the use of FEC to preserve quality under packet-loss conditions.? ................
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