RTP control protocol (RTCP) extended report (XR) block for independent reporting of burst/fgp discard metrics

This document defines an RTP Control Protocol (RTCP) Extended Report (XR) block that allows the reporting of burst/gap discard metrics independently of the burst/gap loss metrics for use in a range of RTP applications

Duplicating RTP streams

Packet loss is undesirable for real-time multimedia sessions but can occur due to a variety of reasons including unplanned network outages. In unicast transmissions, recovering from such an outage can be difficult depending on the outage duration, due to the potentially large number of missing packets. In multicast transmissions, recovery is even more challenging as many receivers could be impacted by the outage. For this challenge, one solution that does not incur unbounded delay is to duplicate the packets and send them in separate redundant streams, provided that the underlying network satisfies certain requirements. This document explains how Real-time Transport Protocol (RTP) streams can be duplicated without breaking RTP or RTP Control Protocol (RTCP) rule

Rtp and the datagram congestion control protocol

We describe how the new Datagram Congestion Control Protocol (DCCP) can be used as a bearer for the Real-time Transport Protocol (RTP) to provide a congestion controlled basis for networked multimedia applications. This is a step towards deployment of congestion control for such applications, necessary to ensure the future stability of the best-effort network if high-bandwidth streaming and IPTV services are to be deployed outside of closed QoS-managed networks

Mobile Multimedia Streaming Library

In recent years, multimedia has become a commonly used tool for presenting contents to the users. The employment of multimedia is no longer limited to only the entertainment industry, but spans in other areas as well. In academics, lectures are recorded to audio and video for storage and distribution to students. Free online multimedia hosting services are popularly cherished, such as “youtube.com” and “yahoo video”, and with the increasing affordability of digital camera, hundreds, or maybe thousands, of home-made videos and music audio are created daily and published online. Low-cost digital recorders such as webcams also help promote the use of video for surveillance, both for commercial and personal use. Suddenly, there comes the need for digital multimedia delivery, which happens naturally with the advancement in Internet bandwidth and the popularity of multimedia sharing. Multimedia delivery comes in two methods: downloading and streaming. Streaming requires more complex structure, but rewards with better user experience. Although streaming is the method of choice today, downloading is still useful in ad-hoc situation where streaming is not feasible. This project aims to provide streaming-like capability to mobile devices. Since mobile gadgets are limited in resources compared to personal computers (PC), streaming sometimes is the only way to deliver media contents to user. This work targets devices in the so-called “ad-hoc situation”, and also seeks to save the cost associated with multimedia streaming, which traditionally uses the operator wireless network, by using a LAN-connected proxy and the Bluetooth medium. It is also to serve the educational purpose in learning about multimedia streaming on cellular phones. This project experiments with several approaches to implement streaming on mobile phones. It discusses each approach in details. Finally, a library and a sample application are implemented to demonstrate the solution

Sending multiple RTP streams in a single RTP session: grouping RTP control protocol (RTCP) reception statistics and other feedback

RTP allows multiple RTP streams to be sent in a single session but requires each Synchronization Source (SSRC) to send RTP Control Protocol (RTCP) reception quality reports for every other SSRC visible in the session. This causes the number of RTCP reception reports to grow with the number of SSRCs, rather than the number of endpoints. In many cases, most of these RTCP reception reports are unnecessary, since all SSRCs of an endpoint are normally co-located and see the same reception quality. This memo defines a Reporting Group extension to RTCP to reduce the reporting overhead in such scenarios

RTP Control Protocol (RTCP) Feedback for Congestion Control

An effective RTP congestion control algorithm requires more fine-grained feedback on packet loss, timing, and Explicit Congestion Notification (ECN) marks than is provided by the standard RTP Control Protocol (RTCP) Sender Report (SR) and Receiver Report (RR) packets. This document describes an RTCP feedback message intended to enable congestion control for interactive real-time traffic using RTP. The feedback message is designed for use with a sender-based congestion control algorithm, in which the receiver of an RTP flow sends back to the sender RTCP feedback packets containing the information the sender needs to perform congestion control

Implementación de servidor SIP (Session initiation protocol)

En este documento se plantea toda la teoría necesaria para el entendimiento del protocolo SIP y absolutamente todos los procesos y protocolos que permiten una comunicación de voz sobre IP entre dos usuarios, cualquiera sea su ubicación. Además, se da una guía completa para implementar un servicio de telefonía en Internet, siguiendo todos los pasos para que se logre un mejoramiento continuo y una evolución de esta teoría, siendo esta el futuro de la telefonía multiusuario. En este documento se exponen también las soluciones prácticas a las limitantes y conflictos que se presentan en distintos escenarios de uso de los servicios de voz sobre IP para lograr la implementación correcta de estos por los interesados en hacerloTrabajo de Grado (Ingeniero Electrónico)-- Universidad Autónoma de Occidente.2004PregradoIngeniero(a) en Electrónica y Telecomunicacione