[Live-devel] How to implement Opus live audio

[Roland Aigner]

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 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 of response data.
	Received a complete DESCRIBE response:
	RTSP/1.0 404 File Not Found, Or In Incorrect Format
	CSeq: 2

Any ideas what this is about?


-----Ursprüngliche Nachricht-----
Von: live-devel [mailto:live-devel-bounces at ns.live555.com] Im Auftrag von Ross Finlayson
Gesendet: Freitag, 2. Juni 2017 10:19
An: LIVE555 Streaming Media - development & use <live-devel at ns.live555.com>
Betreff: Re: [Live-devel] How to implement Opus live audio

> 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 your “RTPSink” (subclass) object, just call:
	SimpleRTPSink::createNew(envir(), rtpGroupsock, rtpPayloadTypeIfDynamic, 48000, "audio", "OPUS", 2, False); (The reason for the final ‘False’ is that only one Opus ‘packet’ is allowed in each RTP packet.)


Ross Finlayson
Live Networks, Inc.
http://www.live555.com/


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