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Scalable Networked Multipoint Multimedia Conferencing and Telepresence
Published in Radhika Ranjan Roy, Handbook on Networked Multipoint Multimedia Conferencing and Multistream Immersive Telepresence using SIP, 2020
Different audio (e.g., International Telegraph Union – Telecommunication [ITU-T] G-series) and video (e.g., Moving Picture Expert Group [MPEG], Joint Photographic Expert Group [JPEG], and ITU-T H-series) codecs are used in multimedia sessions by conference participants. The bit streams of each codec are transferred over the real-time transport protocol (RTP) for transferring over UDP/IP. However, the common codec type either for audio or for video is negotiated by the SIP signaling messages that contain the information for each codec type that is proposed by the conference participants. SIP does not mandate any audio codec for any media that should be used by each participant if no common codec is supported among the conferees. In this situation, the conferees may use the transcoding services for preventing failures of the session establishment. The real-time transport control protocol (RTCP) is based on the periodic transmission of control packets to all participants in the session and provides feedback on the quality of the data (e.g., RTP packets of audio/video) distribution. This is an integral part of the RTP’s role as a transport protocol and is related to the flow and congestion control functions of other transport protocols.
Networked Multimedia Protocols
Published in Radhika Ranjan Roy, Handbook of SDP for Multimedia Session Negotiations, 2018
SIP, being an application layer signaling protocol, facilitates the creation of networked multimedia application services such as teleconferencing (TC), video teleconferencing (VTC)/VC, and multimedia application sharing. Multimedia applications may need to support different audio (e.g., International Telegraph Union – Telecommunication (ITU-T) G-series) and video (e.g., Moving Picture Expert Group, Joint Photographic Expert Group, and ITU-T H-series) codecs in multimedia sessions by conference participants. The bit streams of each codec type are transferred using RTP over the User Datagram Protocol (UDP), Stream Control Transmission Protocol (SCTP) over the Internet Protocol (IP) network. However, RTCP is used for the periodic transmission of control packets to all participants in the session, and it provides feedback on the quality of the data (e.g., RTP packets of audio/video) distribution. This is an integral part of the RTP’s role as a transport protocol and is related to the flow and congestion control functions of other transport protocols. The data (e.g., text, graphics, and still pictures) application may be transferred using Transmission Control Protocol (TCP). Figure 1.1 shows the relationship between SIP and other protocols.
Telepresence: Immersive Experience and Interoperability
Published in Hassnaa Moustafa, Sherali Zeadally, Media Networks: Architectures, Applications, and Standards, 2016
SIP-based systems use the IETF-defined Real-Time Transport Protocol (RTP) to carry audio and video streams between communication endpoints. They also use the Real-Time Transport Control Protocol (RTCP), a sister protocol of RTP aimed at providing feedback on the QoS, by periodically sending statistics information to senders of RTP traffic. Beside basic feedback, video communications, including TP, use RTCP for reporting lost pictures and for sending codec control messages such as the Full Intra Request (FIR) command—also known as “video fast update request” defined in RFC5104 [13]—which requests the media source to send back a decoder refresh point to reset the decoder to a known state. In the context of TP, RTP is used with either the Audio/Visual Profile (AVP) defined in RFC3551 [14] or preferably the early feedback profile (AVPF) defined in RFC4585 [15]. Using one profile or the other will determine the type of feedback that a receiver can send using RTCP. Other profiles may be used as well if media flow encryption is required.
Efficient video transmission—a critical review of various protocols and strategies
Published in Journal of the Chinese Institute of Engineers, 2021
Ali Siddique, Abdul Rauf Bhatti, Ahmed Bilal Awan, Arslan Dawood Butt, Ali S. Alghamdi, Muhammad Farhan, Nadia Rasheed
The Real-time transport protocols (RTP) and HTTP are the two commonly used application-layer protocols. Although RTP is not an application-level protocol, it is often compared with HTTP. The RTP is normally run over UDP and may be easily used with SCTP or DCCP, whereas HTTP generally runs over TCP. For more details on how RTP is used with DCCP, refer to Perkins (2010). If the RTP is to be used over TCP, a framing mechanism is required (Lazzaro 2006). The HTTP is typically used for pre-coded video streaming applications. The RTP is accompanied by a control protocol i.e. real-time control protocol (RTCP, see [Schulzrinne et al. 2003]). The RTCP can provide quality-related feedback information for an ongoing RTP session, together with identification information for participants of an RTP session. The RTCP monitors transmission statistics and QoS information. The RTCP can inform the encoder about the network and transmission path characteristics as observed by the decoder. The encoder may then adapt to the error-resilience and transmission rates in real-time.
An investigation on adaptive HTTP media streaming Quality-of-Experience (QoE) and agility using cloud media services
Published in International Journal of Computers and Applications, 2021
Selvaraj Kesavan, E. Saravana Kumar, Abhishek Kumar, K. Vengatesan
Streaming is the process of dividing data in the file is bro-ken into small packets that are sent in a steady and continuous flow, as a stream to the end device. As soon as few initial data packets received, the playback starts as the rest of the packets are transferred to the end user's device while playing. The client plays out buffer makes sure that the playback to continue uninterrupted despite variations in the rate of received rate and network delay. RTP over UDP widely used in low latency media and entertainment applications such as streaming, video telephony, video conference, set top box application and push-to-talk features. RTP and RTCP protocols are used for payload transmission and control, respectively. Generally Real Time Streaming Protocol (RTSP) over TCP is used for session initiation and description even though specification allows RTSP over UDP.