Re: [Live-devel] OPUS encoded stream via SimpleRTPSink

One more thing: I recommend that you first test your server using “openRTSP” as your RTSP client, rather than immediately trying to use a media player (such as VLC). Run “openRTSP” on your “rtsp://“ URL. Make sure that you get a non-zero-sized audio file (it

Re: [Live-devel] OPUS encoded stream via SimpleRTPSink

> I did not fully get the part about the RTP packet assembly. The RTP > packet (12-byte header) is created by the RTPSink, right? Yes, we fill this in automatically. All you need to provide is the payload (in this case, one OPUS ‘packet’ each time). > So after > digging through the RFCs and the

Re: [Live-devel] OPUS encoded stream via SimpleRTPSink

> You’re correct that “SimpleRTPSink” is the correct ‘sink’ class to > use. (You can do this because the RTP payload format for OPUS audio > - defined in RFC 7587 - is relatively straightforward.) > Note that - from RFC 7587, section 4.2 - a RTP packet contains > exactly one ‘OPUS packet’, which