After enough packet received to increase rtp packetization time (ptime) the we forward the message over raw socket and set dscp to be 10 so that this time packets can escape iptable rules. In our app we do not forward packet immediately. And it does not drop packets or anything. Then insert the packets into a new user managed queue and does some transformations on it, like concatenation of udp payload. The application receives packets from kernel using netfilter_queue library. We are getting very poor quality of voice during testing of a new filtering application of us. To: CentOS mailing list RTP stats explaination It seems you have some timing issues with your app. Go to the packets with high "Delta" and verify that you have gaps in your RTP stream. "Max delta" should not be very different from the RTP packetization time. If not, then the packets of your stream should be evenly spaced over time, i.e. To: Community support list for Wireshark, ", ", CentOS mailing list, " Subject: RE: RTP stats explainationĪre you using some sort of silence suppression (or voice activity detection) ?.
0 kommentar(er)
