基于RTP的IP实时音频传输研究

On Developing IP Real Time Audio Transport System Based on RTP for Conferencing in China

实时传输协议 (RTP)是当前 Internet上较理想的多媒体通信协议 ,从应用的角度出发 ,简要介绍了 RTP协议适合实时传输的特性及其数据包和控制包的格式 ,并着重讨论了如何利用该协议来满足实时音频传输的高实时性要求。在开发的 IP会议系统中 ,经过滤波处理的音频数据被合理的分割为网络数据单元 ,然后组装成 RTP数据包进行传输 ;在通讯过程的控制方面 ,系统周期性的组播包含各个会议站点信息 SDES包 ,同时各接收站点要向组内组播包含其接收的业务质量情况的 Qo S包。实验表明 ,该系统在 Internet和 Intranet上运行情况良好。

We used realtime transport protocol (RTP) to satisfy the strict realtime requirements of the conferencing system developed by us. First the filtered audio data was appropriately divided so as to form a network data unit (NDU), then the NDU was encapsulated in RTP data package for transporting. This step was schematically shown in Fig.2 and discussed in much detail in section 2. For control of the communication process, we used mainly two kinds of RTCP (realtime transport control protocol) packages: SDES (source description) and QoS (quality of service). SDES package contains information about all the conference participants. The conferencing system multicasts SDES package periodically. QoS package contains the quality status of the receiving transaction at one of the participant sites. Each participant site multicasts its QoS package periodically. Our tests show preliminarily that the conferencing system developed by us works well on both Internet and Intranet.