[Live-devel] Setting BasicUDPSource in RTSPClient code to automatically request UDP.

? Regards Taha Mohammad Taha; Sr. Software Developer; Wow Vision India Pvt. Ltd; ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel

Re: [Live-devel] "MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size in Using MPEG-2 Transport Stream, coming from a live UDP (raw-UDP or RTP/UDP)

ut file is not valid. Kindly suggest me, if i am doing something wrong. Please find attachment. Regards Taha Mohammad Taha; Sr. Software Developer; Wow Vision India Pvt. Ltd; On 2017-01-06 15:54, Ross Finlayson wrote: Yes i am using “mpeg2TransportStreamFromUDPSourceTest” example code(i.e.,

Re: [Live-devel] "MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size in Using MPEG-2 Transport Stream, coming from a live UDP (raw-UDP or RTP/UDP)

displayed and no data is displayed on VLC with RTSP : "rtsp://192.168.100.217:8554/mpeg2Ts". Regards Taha Mohammad Taha; Sr. Software Developer; Wow Vision India Pvt. Ltd; On 2017-01-06 13:55, Ross Finlayson wrote: Are you using the “mpeg2TransportStreamFromUDPSourceTest” example

[Live-devel] "MultiFramedRTPSource::doGetNextFrame1(): The total received frame size exceeds the client's buffer size in Using MPEG-2 Transport Stream, coming from a live UDP (raw-UDP or RTP/UDP) sour

size exceeds the client's buffer size(--). bytes of trailing data will be dropped! Kindly help me in fixing this. Regards Taha Mohammad Taha; Sr. Software Developer; Wow Vision India Pvt. Ltd; ___ live-devel mailing list live-devel@lists.l

Re: [Live-devel] Streaming to RTSP clients with different ports

penRTSP/ <http://live555.com/openRTSP/> >>Ross Finlayson >>Live Networks, Inc. >>http://www.live555.com/ Thanks, Taha ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel

[Live-devel] Streaming to RTSP clients with different ports

Hi, I have a test scenario, where I am streaming webcam input + mic over to a remote server using RTP protocol, which re-streams this to another client (other client pulls data through RTSP): FFmpeg (RTP AAC audio + RTP video) -> Server -> (RTSP AAC audio + RTSP video) FFmpeg client In