> However, when I connect with my client via an RTSPClient, I get a 404
> returned.
>
> Sending request: DESCRIBE rtsp://192.168.5.144/audio RTSP/1.0
> CSeq: 2
> User-Agent: LIVE555 Streaming Media v2016.11.28
> Accept: application/sdp
>
> Received 101 new bytes o
Thanks, that did take me a tiny step further. I'm now creating an RTPSink in
createNewRTPSink just like you suggested, and an instance of my custom Opus
encoder source in createNewStreamSource. However, when I connect with my client
via an RTSPClient, I get a 404 returned.
Sending requ
hi Ross,
see below for the explanation from Frederik of dereferencing a deleted
RTPInterface
-
When I take the following steps to add debug output:
* #define DEBUG_SEND in RTPInterface.cpp
* print RTPInterface's address in its destructor
* print which RTPInterface address is used in Sock
> I'm trying to stream Opus-encoded live audio via live555 from my server and
> I'm a bit lost in how to implement that.
Fortunately the RTP payload format for Opus audio is very simple - so you can
use the existing “SimpleRTPSink” class for this - without modification.
When you are creating yo
Hi,
I'm trying to stream Opus-encoded live audio via live555 from my server and I'm
a bit lost in how to implement that. I was trying to orient on live555 sample
code and on how streaming of OGG files is done, but I'm losing track in the
code at some point. Is there a sample I can use as kind o
