Virtual RTCP: A Case Study of Monitoring and Repair for UDP-based IPTV Systems

IPTV systems have seen widespread deployment, but often lack robust mechanisms for monitoring the quality of experience. This makes it difficult for network operators to ensure that their services match the quality of traditional broadcast TV systems, leading to consumer dissatisfaction. We present a case study of virtual RTCP, a new framework for reception quality monitoring and reporting for UDP-encapsulated MPEG video delivered over IP multicast. We show that this allows incremental deployment of reporting infrastructure, coupled with effective retransmission-based packet loss repair

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 MIDI : Recovery Journal Evaluation and Alternative Proposal

An RTP payload for MIDI commands is under development. As a part of this draft, a default resiliency mechanism for the transport over lossy networks defines a journalling method called recovery journal. But the theoretical size of this recovery journal can be very large and its format is complex. This report will present an empirical evaluation of the recovery journal size based on a few MidiFiles. We will also propose an alternative solution for the resiliency of RTP MIDI streams based on the combined use of redundancy and retransmissions. Our solution is simpler and might be interesting for some scenarios, typically: short grouping times, complex streams or unconventional semantics

Real-time Audio-Visual Media Transport over QUIC

We consider the problem of how to transport low-latency, interactive, real-time traffic over QUIC. This is needed to support applications like WebRTC, but difficult to support due to the reliable, unframed, nature of QUIC streams. We review the needs of low-latency real-time applications and how they have been supported in previous protocols, then propose a minimal set of extensions to QUIC to provide such support. Compared to a raw datagram service, our extensions provide meaningful support for partially reliable and real-time flows, in a backwards compatible manner

Providing End-to-End Connectivity to SIP User Agents Behind NATs

The widespread diffusion of private networks in SOHO scenarios is fostering an increased deployment of Network Address Translators (NATs). The presence of NATs seriously limits end-to-end connectivity and prevents protocols like the Session Initiation Protocol (SIP) from working properly. This document shows how the Address List Extension (ALEX), which was originally developed to provide dual-stack and multi-homing support to SIP, can be used, with minor modifications, to ensure end-to-end connectivity for both media and signaling flows, without relying on intermediate relay nodes whenever it is possibl