The Experts below are selected from a list of 69 Experts worldwide ranked by ideXlab platform

Shiquan Xiao - One of the best experts on this subject based on the ideXlab platform.

  • Design and Implementation of Service Control Server in IMS-Based IPTV
    2010 International Conference on Internet Technology and Applications, 2010
    Co-Authors: Dong Yu, Shiquan Xiao
    Abstract:

    IMS-based IPTV is an important research direction in next generation IPTV, in which Service Control Server (SCS) as an IMS application Server is one of the most critical network entities. By the comparison of differences in behavior of various SIP entities, such as Proxy Server, Virtual User Agent (VUA), Back-to-Back User Agent and Redirect Server, combined with the characteristic of different IPTV services, such as Broadcast, Content on Demand (CoD), Time-shift Television, and network-side Private Video Record, some suggestions have been given on how to choose the SIP entities for SCS. After that, a general layered architecture for SCS has been designed, which can be used for different SIP entities, especially a modification has been made for VUA with the signaling and media function integration. Based on the analysis of SIP signaling and RTSP media control flows, the service state machine on the top layer in SCS has been designed through the mapping relationship of session event and service logical states. Further more, taking the CoD service as an example, we implement a SCS prototype and the technical feasibility has been verified.

Rakhmadhany Primananda - One of the best experts on this subject based on the ideXlab platform.

  • Implementation of The SIP Express Router with Mediaproxy on VoIP
    International journal of advanced computer science, 2012
    Co-Authors: Heru Nurwarsito, R. Arief Setyawan, Rakhmadhany Primananda
    Abstract:

    Voice Over IP (VoIP) is a technology that could pass the voice traffic and data packet form over an IP network. Network can be used for intranet or Internet. Phone calls using VoIP has advantages in terms of cheaper cost of PSTN phone to more than half, because the cost is calculated by the cost of the global nature of the Internet. Session Initiation Protocol (SIP) is a signaling protocol at the application layer that serves to establish, modify, and terminate a multimedia session involving one or more users. This SIP signaling has SIP message in text form that is used for session management by the SIP components, such as User Agent, Registrar, Redirect Server, and Proxy Server. Building a SIP communication requires SIP Express Router (SER) to be able to receive SIP messages, for handling the basic functions of SIP messages. Problems occur when the NAT through which the voice communication affected, blocked, starting from the sound that is not sent or one side of the sound are sent (half duplex). The method that could be used to penetrate NAT is to use a given mediaproxy random RTP port to penetrate NAT.

  • Implementation of the SIP Express Router with Mediaproxy Method on VoIP
    World Academy of Science Engineering and Technology International Journal of Electrical Computer Energetic Electronic and Communication Engineering, 2010
    Co-Authors: Heru Nurwarsito, R. Arief Setyawan, Rakhmadhany Primananda
    Abstract:

    Voice Over IP (VoIP) is a technology that could pass the voice traffic and data packet form over an IP network. Network can be used for intranet or Internet. Phone calls using VoIP has advantages in terms of cheaper cost of PSTN phone to more than half, because the cost is calculated by the cost of the global nature of the Internet. Session Initiation Protocol (SIP) is a signaling protocol at the application layer that serves to establish, modify, and terminate a multimedia session involving one or more users. This SIP signaling has SIP message in text form that is used for session management by the SIP components, such as User Agent, Registrar, Redirect Server, and Proxy Server. Building a SIP communication requires SIP Express Router (SER) to be able to receive SIP messages, for handling the basic functions of SIP messages. Problems occur when the NAT through which the voice communication affected, blocked, starting from the sound that is not sent or one side of the sound are sent (half duplex). The method that could be used to penetrate NAT is to use a given mediaproxy random RTP port to penetrate NAT.

Dong Yu - One of the best experts on this subject based on the ideXlab platform.

  • Design and Implementation of Service Control Server in IMS-Based IPTV
    2010 International Conference on Internet Technology and Applications, 2010
    Co-Authors: Dong Yu, Shiquan Xiao
    Abstract:

    IMS-based IPTV is an important research direction in next generation IPTV, in which Service Control Server (SCS) as an IMS application Server is one of the most critical network entities. By the comparison of differences in behavior of various SIP entities, such as Proxy Server, Virtual User Agent (VUA), Back-to-Back User Agent and Redirect Server, combined with the characteristic of different IPTV services, such as Broadcast, Content on Demand (CoD), Time-shift Television, and network-side Private Video Record, some suggestions have been given on how to choose the SIP entities for SCS. After that, a general layered architecture for SCS has been designed, which can be used for different SIP entities, especially a modification has been made for VUA with the signaling and media function integration. Based on the analysis of SIP signaling and RTSP media control flows, the service state machine on the top layer in SCS has been designed through the mapping relationship of session event and service logical states. Further more, taking the CoD service as an example, we implement a SCS prototype and the technical feasibility has been verified.

Heru Nurwarsito - One of the best experts on this subject based on the ideXlab platform.

  • Implementation of The SIP Express Router with Mediaproxy on VoIP
    International journal of advanced computer science, 2012
    Co-Authors: Heru Nurwarsito, R. Arief Setyawan, Rakhmadhany Primananda
    Abstract:

    Voice Over IP (VoIP) is a technology that could pass the voice traffic and data packet form over an IP network. Network can be used for intranet or Internet. Phone calls using VoIP has advantages in terms of cheaper cost of PSTN phone to more than half, because the cost is calculated by the cost of the global nature of the Internet. Session Initiation Protocol (SIP) is a signaling protocol at the application layer that serves to establish, modify, and terminate a multimedia session involving one or more users. This SIP signaling has SIP message in text form that is used for session management by the SIP components, such as User Agent, Registrar, Redirect Server, and Proxy Server. Building a SIP communication requires SIP Express Router (SER) to be able to receive SIP messages, for handling the basic functions of SIP messages. Problems occur when the NAT through which the voice communication affected, blocked, starting from the sound that is not sent or one side of the sound are sent (half duplex). The method that could be used to penetrate NAT is to use a given mediaproxy random RTP port to penetrate NAT.

  • Implementation of the SIP Express Router with Mediaproxy Method on VoIP
    World Academy of Science Engineering and Technology International Journal of Electrical Computer Energetic Electronic and Communication Engineering, 2010
    Co-Authors: Heru Nurwarsito, R. Arief Setyawan, Rakhmadhany Primananda
    Abstract:

    Voice Over IP (VoIP) is a technology that could pass the voice traffic and data packet form over an IP network. Network can be used for intranet or Internet. Phone calls using VoIP has advantages in terms of cheaper cost of PSTN phone to more than half, because the cost is calculated by the cost of the global nature of the Internet. Session Initiation Protocol (SIP) is a signaling protocol at the application layer that serves to establish, modify, and terminate a multimedia session involving one or more users. This SIP signaling has SIP message in text form that is used for session management by the SIP components, such as User Agent, Registrar, Redirect Server, and Proxy Server. Building a SIP communication requires SIP Express Router (SER) to be able to receive SIP messages, for handling the basic functions of SIP messages. Problems occur when the NAT through which the voice communication affected, blocked, starting from the sound that is not sent or one side of the sound are sent (half duplex). The method that could be used to penetrate NAT is to use a given mediaproxy random RTP port to penetrate NAT.

R. Arief Setyawan - One of the best experts on this subject based on the ideXlab platform.

  • Implementation of The SIP Express Router with Mediaproxy on VoIP
    International journal of advanced computer science, 2012
    Co-Authors: Heru Nurwarsito, R. Arief Setyawan, Rakhmadhany Primananda
    Abstract:

    Voice Over IP (VoIP) is a technology that could pass the voice traffic and data packet form over an IP network. Network can be used for intranet or Internet. Phone calls using VoIP has advantages in terms of cheaper cost of PSTN phone to more than half, because the cost is calculated by the cost of the global nature of the Internet. Session Initiation Protocol (SIP) is a signaling protocol at the application layer that serves to establish, modify, and terminate a multimedia session involving one or more users. This SIP signaling has SIP message in text form that is used for session management by the SIP components, such as User Agent, Registrar, Redirect Server, and Proxy Server. Building a SIP communication requires SIP Express Router (SER) to be able to receive SIP messages, for handling the basic functions of SIP messages. Problems occur when the NAT through which the voice communication affected, blocked, starting from the sound that is not sent or one side of the sound are sent (half duplex). The method that could be used to penetrate NAT is to use a given mediaproxy random RTP port to penetrate NAT.

  • Implementation of the SIP Express Router with Mediaproxy Method on VoIP
    World Academy of Science Engineering and Technology International Journal of Electrical Computer Energetic Electronic and Communication Engineering, 2010
    Co-Authors: Heru Nurwarsito, R. Arief Setyawan, Rakhmadhany Primananda
    Abstract:

    Voice Over IP (VoIP) is a technology that could pass the voice traffic and data packet form over an IP network. Network can be used for intranet or Internet. Phone calls using VoIP has advantages in terms of cheaper cost of PSTN phone to more than half, because the cost is calculated by the cost of the global nature of the Internet. Session Initiation Protocol (SIP) is a signaling protocol at the application layer that serves to establish, modify, and terminate a multimedia session involving one or more users. This SIP signaling has SIP message in text form that is used for session management by the SIP components, such as User Agent, Registrar, Redirect Server, and Proxy Server. Building a SIP communication requires SIP Express Router (SER) to be able to receive SIP messages, for handling the basic functions of SIP messages. Problems occur when the NAT through which the voice communication affected, blocked, starting from the sound that is not sent or one side of the sound are sent (half duplex). The method that could be used to penetrate NAT is to use a given mediaproxy random RTP port to penetrate NAT.