Nowadays portable devices with multiple wireless interfaces and using multimedia services are becoming more popular on the Internet. This paper describes a family of multipath binomial congestion control algorithms for audio/video streaming, where a low variant of transmission rate is important. We extend the fluid model of binomial algorithms for single-path transmission to support the concurrent transmission of packets across multiple paths. We focus on the extension of two particular algorithms, SQRT and IIAD, for multiple paths, called MPSQRT and MPIIAD, respectively. Additionally, we apply the design technique (using the multipath fluid model) for multipath TCP (MPTCP) into the extension of SQRT and IIAD, called fbMPSQRT and fbMPIIAD, respectively. Both two approaches ensure that multipath binomial congestion control algorithms achieve load-balancing, throughput improvement, and fairness to single-path binomial algorithms at shared bottlenecks. Through the simulations and comparison with the uncoordinated protocols MPSQRT/MPIIAD, fbMPSQRT/fbMPIIAD and MPTCP, we find that our extended multipath transport protocols can preserve lower latency and transmission rate variance than MPTCP, fairly share with single-path SQRT/IIAD, MPTCP and TCP, and also can achieve throughput improvements and load-balancing equivalent to those of MPTCP under various scenarios and network conditions.

We propose a parametric packet-layer model for monitoring audio quality in multimedia streaming services such as Internet protocol television (IPTV). This model estimates audio quality of experience (QoE) on the basis of quality degradation due to coding and packet loss of an audio sequence. The input parameters of this model are audio bit rate, sampling rate, frame length, packet-loss frequency, and average burst length. Audio bit rate, packet-loss frequency, and average burst length are calculated from header information in received IP packets. For sampling rate, frame length, and audio codec type, the values or the names used in monitored services are input into this model directly. We performed a subjective listening test to examine the relationships between these input parameters and perceived audio quality. The codec used in this test was the Advanced Audio Codec-Low Complexity (AAC-LC), which is one of the international standards for audio coding. On the basis of the test results, we developed an audio quality evaluation model. The verification results indicate that audio quality estimated by the proposed model has a high correlation with perceived audio quality.

One of the important problems in overlay multicast is how to deal with node failures and ungraceful leavings. When a non-leaf end host fails or leaves the multicast session, all downstream nodes will be affected. In this paper, we adopt the proactive approach, which pre-calculates a candidate node (called parent-to-be) for each node to connect to in case its current parent dies. The goal is to recover the overlay multicast tree quickly so that the disruption of service to those affected nodes is minimized. We combine the local parent-to-be locating and global parent-to-be locating schemes together, in order to take advantage of less interference in the local scheme and the flexibility of the global scheme. The quality of the recovered tree is improved while the responsiveness of the proactive approach is maintained.

In this paper, we propose a set of broadcast streaming protocols designed for unidirectional radio channels. Considering the limited size and implementation overhead on a mobile terminal, the proposed protocol set is almost compliant with the current mobile streaming protocols, i.e. 3GPP PSS (Packet-switched Streaming Service), except for that the proposed protocols are designed to work on a unidirectional downlink channel. This protocol set enables flexible layout rendering by SMIL (Synchronized Multimedia Integration Language) in combination with SDP (Session Description Protocol), and reliable and synchronized static media (including still image and text) delivery by RTP (Real-time Transport Protocol) carousel. We present the prototype of this protocol set and measure its performance of video quality and waiting time for video presentation through a W-CDMA radio channel emulator and header compression nodes. From the experimental results, we show 1) trade-off between video quality and waiting time, 2) advantage and disadvantage of header compression, 3) effectiveness of synchronized transmission of SDP, SMIL, and I-frames of video objects, and 4) reliability of RTP-carousel. This protocol set is applicable to 3G MBMS (Multimedia Broadcast/Multicast Service) streaming service.