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Riga 136: | Riga 136: | ||
The Impact of Heterogeneous Bandwidth Constraints on DHT-Based Multicast | The Impact of Heterogeneous Bandwidth Constraints on DHT-Based Multicast | ||
</paper> | </paper> | ||
− | <paper authors="Sripanidkulchai, K. and Ganjam, A. and Maggs, B. and Zhang, H." conference="Proceedings of the 2004 conference on Applications, technologies, architectures, and protocols for computer communications" date="2004 ([http://msw3.stanford.edu/~ivmsjc/esetton/jc_feb16.pdf PDF])" place="ACM Press New York, NY, USA" > | + | <paper authors="Sripanidkulchai, K. and Ganjam, A. and Maggs, B. and Zhang, H." conference="Proceedings of the SIGCOMM 2004 conference on Applications, technologies, architectures, and protocols for computer communications" date="2004 ([http://msw3.stanford.edu/~ivmsjc/esetton/jc_feb16.pdf PDF])" place="ACM Press New York, NY, USA" > |
The feasibility of supporting large-scale live streaming applications with dynamic application end-points | The feasibility of supporting large-scale live streaming applications with dynamic application end-points | ||
</paper> | </paper> |
The main goal of this research effort is designing and implementing the technologies that will enable efficient and scalable audio/video distribution over IP. In order to reach such goal several fundamental building blocks have to be carefully designed and implemented:
The figure shown below depicts the fundamental building blocks of the multimedia distribution system.
The end to end transport protocol is a key component of the system. We address the problem of congestion control of multimedia contents which is still a very active research topic. The ever increasing amount of data traffic due to VoIP and video on demand applications poses a potential challenge to the stability of the Internet, that has been guaranteed by the TCP congestion control developed by V. Jacobson. Applications as Skype, RealNetwork don't employ a responsive congestion control algorithm and they do not compete fairly with TCP flows (see this page)
Multimedia congestion control differs from bulk data congestion protocols in several aspects:
Several congestion control algorithms for multimedia transport have been proposed in the last years such as TEAR and TFRC (TCP Friendly Rate Control), and recently DCCP which is a kernel space implementation of TFRC.
However it has been shown in recent literature [Balan] that TFRC (even using the small packet version) does not perform well when used in VoIP scenarios, achieving worse performance with respect to TCP.
Our purpose is to design and implement in user space a protocol a-la Westwood+ which will be able to scale from low bandwidth links (few Kbps such as GSM) to very high bandwidth links (such as gigabit networks).
As we aim at seamless multimedia delivery over a wide range of link capacities and technologies (wired and wireless), the transport protocol must guarantee its functioning even in the case of lossy links.
Moreover we aim at integrating the congestion control algorithm with a quality adaptation layer which is in charge of adapting the quality in the underlying encoder in order to get the maximum audio/video quality without congesting the network and losing packets.
The task of the Quality Adaptation Layer is to serve as a communication layer between Adaptive Codecs and the E2E Transport protocol.
The estimated available bandwidth, RTT, Jitter and Loss Rate are the input of this sub-block, which are passed by the E2E protocol. Based on the charateristics of the codecs this module is able to set the correct parameters in the scalable codecs in order to adapt the quality to match the available bandwidth and pursue congestion avoidance and packet loss.
Each supported codec must have a "Codec Controller" which takes the feedback of the E2E protocol in order to evaluate the quality parameter(s) to be set in the codec and pursue quality adaptation.
Streaming of live contents requires low packet latency. This can be accomplished reducing buffer levels on sender, so the audio/video encoding level should be adapted to the transport sending rate.
Adaptive audio and video codecs can accomplish this task. Some audio/video codecs used in our researches are the following:
Video scalability can be assured by using two different techniques:
We use the GStreamer development platform for the implementation of test client/server applications.
Nowadays the popularity of multimedia content distribution over Internet is increasing. Providing live stream or video on demand services over Internet is a challenging problem for providers because:
In order to reduce the bandwidth requirements for content providers, the best solution is using End System Multicasting over Peer-To-Peer (P2P) networks.
End System Multicasting is a solution for one-to-many and many-to-many distribution services. Classical (IP) Multicast is based on minimal functionalities offered by IP level and it lacks of flow and congestion control, session management, and so on. End System Multicasting implements advanced distribution services at application level, so that each node being involved in the distribution should forward the stream to several other nodes using an unicast protocol (TCP or UDP). P2P topologies would help reducing this flow duplication.
The presentation (Overlay Networks) describes the state of art of IP Multicasting. The main issues involved in this research field are the following:
The main goal of this research effort is designing and implementing the technologies that will enable efficient and scalable audio/video distribution over IP. In order to reach such goal several fundamental building blocks have to be carefully designed and implemented:
The figure shown below depicts the fundamental building blocks of the multimedia distribution system.
The end to end transport protocol is a key component of the system. We address the problem of congestion control of multimedia contents which is still a very active research topic. The ever increasing amount of data traffic due to VoIP and video on demand applications poses a potential challenge to the stability of the Internet, that has been guaranteed by the TCP congestion control developed by V. Jacobson. Applications as Skype, RealNetwork don't employ a responsive congestion control algorithm and they do not compete fairly with TCP flows (see this page)
Multimedia congestion control differs from bulk data congestion protocols in several aspects:
Several congestion control algorithms for multimedia transport have been proposed in the last years such as TEAR and TFRC (TCP Friendly Rate Control), and recently DCCP which is a kernel space implementation of TFRC.
However it has been shown in recent literature [Balan] that TFRC (even using the small packet version) does not perform well when used in VoIP scenarios, achieving worse performance with respect to TCP.
Our purpose is to design and implement in user space a protocol a-la Westwood+ which will be able to scale from low bandwidth links (few Kbps such as GSM) to very high bandwidth links (such as gigabit networks).
As we aim at seamless multimedia delivery over a wide range of link capacities and technologies (wired and wireless), the transport protocol must guarantee its functioning even in the case of lossy links.
Moreover we aim at integrating the congestion control algorithm with a quality adaptation layer which is in charge of adapting the quality in the underlying encoder in order to get the maximum audio/video quality without congesting the network and losing packets.
The task of the Quality Adaptation Layer is to serve as a communication layer between Adaptive Codecs and the E2E Transport protocol.
The estimated available bandwidth, RTT, Jitter and Loss Rate are the input of this sub-block, which are passed by the E2E protocol. Based on the charateristics of the codecs this module is able to set the correct parameters in the scalable codecs in order to adapt the quality to match the available bandwidth and pursue congestion avoidance and packet loss.
Each supported codec must have a "Codec Controller" which takes the feedback of the E2E protocol in order to evaluate the quality parameter(s) to be set in the codec and pursue quality adaptation.
Streaming of live contents requires low packet latency. This can be accomplished reducing buffer levels on sender, so the audio/video encoding level should be adapted to the transport sending rate.
Adaptive audio and video codecs can accomplish this task. Some audio/video codecs used in our researches are the following:
Video scalability can be assured by using two different techniques:
We use the GStreamer development platform for the implementation of test client/server applications.
Nowadays the popularity of multimedia content distribution over Internet is increasing. Providing live stream or video on demand services over Internet is a challenging problem for providers because:
In order to reduce the bandwidth requirements for content providers, the best solution is using End System Multicasting over Peer-To-Peer (P2P) networks.
End System Multicasting is a solution for one-to-many and many-to-many distribution services. Classical (IP) Multicast is based on minimal functionalities offered by IP level and it lacks of flow and congestion control, session management, and so on. End System Multicasting implements advanced distribution services at application level, so that each node being involved in the distribution should forward the stream to several other nodes using an unicast protocol (TCP or UDP). P2P topologies would help reducing this flow duplication.
The presentation (Overlay Networks) describes the state of art of IP Multicasting. The main issues involved in this research field are the following: