Sending Multiple Types of Media in a Single RTP Session

This document specifies how an RTP session can contain RTP streams with media from multiple media types such as audio, video, and text. This has been restricted by the RTP specifications (RFCs 3550 and 3551), and thus this document updates RFCs 3550 and 3551 to enable this behaviour for applications that satisfy the applicability for using multiple media types in a single RTP session

Analyzing Voice And Video Call Service Performance Over A Local Area Network

Tez (Yüksek Lisans) -- İstanbul Teknik Üniversitesi, Fen Bilimleri Enstitüsü, 2010Thesis (M.Sc.) -- İstanbul Technical University, Institute of Science and Technology, 2010Bu çalışmada, VOIP teknolojisinden ve bu teknolojiyi kablolu ve kablosuz ortamda gerçeklemenin en önemli darboğazları anlatılacaktır. Ayrıca H.323, SIP (Session Initiation Protocol), Megaco ve MGCP gibi yaygın olarak kullanılan ses iletim protokolleri ve H.261, H.263 ve H.264 gibi görüntü iletim protokollerinden bahsedilmiştir. Ses kodek seçimi ve VOIP servis kalitesine etki eden faktörleri anlatılmaktadır. Bu tezde, ses, görüntü ve veri iletişimini aynı anda bünyesinde barındıran gerçek şebekeler simüle edilecektir. Kullanıcılara rastlantısal olarak ses, görüntü ve FTP gibi birtakım uygulamalar atanmıştır. Ayrıca önerilen kablolu şebekeye, kablosuz bir şebeke ilave edilerek sonuçlar incelenecektir. Optimal servis kalitesini sağlamak için seçilen uygun kuyruklama mekanizmaları ve kodek seçimlerini içeren senaryolar incelenecek ve OPNET ile elde edilmiş simülasyon sonuçları tartışılacaktır.In this study, we present a detailed description of the VoIP and also the most common challenges of implementing voice communication into wireline or wireless networks are discussed. Common voice protocols, such as H.323, Session Initiation Protocol (SIP), Megaco, MGCP and video protocols such as H.261, H.263, H.264 are described as well. CODEC selection and factors affecting VoIP Quality of Service are analyzed. We simulate a real network which includes both voice, video and data communication simultaneously. Workstations are randomly assigned to different applications, such as voice, video and FTP. We will also implement a wireless network to our proposed system. The scenarios including selecting appropriate queuing scheme and codec selection are presented and the simulation results with OPNET are drawn.Yüksek LisansM.Sc

Analysing the characteristics of VoIP traffic

In this study, the characteristics of VoIP traffic in a deployed Cisco VoIP phone system and a SIP based soft phone system are analysed. Traffic was captured in a soft phone system, through which elementary understanding about a VoIP system was obtained and experimental setup was validated. An advanced experiment was performed in a deployed Cisco VoIP system in the department of Computer Science at the University of Saskatchewan. Three months of traffic trace was collected beginning October 2006, recording address and protocol information for every packet sent and received on the Cisco VoIP network. The trace was analysed to find out the features of Cisco VoIP system and the findings were presented.This work appears to be one of the first real deployment studies of VoIP that does not rely on artificial traffic. The experimental data provided in this study is useful for design and modeling of such systems, from which more useful predictive models can be generated. The analysis method used in this research can be used for developing synthetic workload models. A clear understanding of usage patterns in a real VoIP network is important for network deployment and potential network activities such as integration, optimizations or expansion. The major factors affecting VoIP quality such as delay, jitter and loss were also measured and simulated in this study, which will be helpful in an advanced VoIP quality study. A traffic generator was developed to generate various simulated VoIP traffic. The data used to provide the traffic model parameters was chosen from peak traffic periods in the captured data from University of Saskatchewan deployment. By utilizing the Traffic Trace function in ns2, the simulated VoIP traffic was fed into ns2, and delay, jitter and packet loss were calculated for different scenarios. Two simulation experiments were performed. The first experiment simulated the traffic of multiple calls running on a backbone link. The second experiment simulated a real network environment with different traffic load patterns. It is significant for network expansion and integration

Quality of services adaptation model for distributed multimedia application based on RTP protocol on IP network

Mémoire numérisé par la Direction des bibliothèques de l'Université de Montréal