基于自适应变速率编码的VoIP网关的研究与应用

针对当前VoIP语音质量易受网络带宽影响的问题，提出了一种基于自适应变速率编码的VoIP网关。通过实时监测RTP语音分组的丢包率，并分析服务质量，自适应地选择最适合当前网络状况的编码速率，以提供一种语音质量和网络状况的最佳组合，从而降低语音分组的丢包率、有效地保障语音服务质量。结合开源项目Asterisk，实现了基于自适应变速率编码的VoIP网关。

Research and application of VoIP gateway based on adaptive variable rate speech coding

In order to solve the problem that the voice of QoS is easily influenced by network bandwidth, a new VolP gateway based on adaptive variable-rate ceding was presented. By real-time computing the ratio of RTP voice packet losing and analyzing the quality of service, it would adaptively choose the most appropriate coding in order to obtain the best combination of voice service and bandwidth, so as to reduce the ratio of voice packet-losing and ensure the quality of voice service. The key techniques such as computing the ratio of RTP voice packet-losing and adjusting bandwidth were discussed intentionally. In the end, the solution has already been applied in the open source program Asterisk.