[Live-devel] RTP Audio without sip

[Ross Finlayson]

> On Jun 13, 2024, at 6:24 AM, Guillermo Bernaldo de Quiros Maraver <gbernaldo at cestel.es> wrote:
> 
> Hi Ross! 
> 
> First of all, thank you so much for your answer! 
> 
> Answering your question, the audio codec is AMR-WB and I have the SDP description too although I know in advance all info related to the media (sample rate, channel number, ...).
> 
> Is there any sample code I can use as a guide?

Yes, I suggest using the code for the “testMP3Receiver” and “testMP3Streamer” applications (both are built automatically in the “testProgs” directory) as a guide.  I suggest starting with the ‘receiver’ application, and make sure that you can receive the incoming AMR-WB audio OK, before working on the ‘streamer’ (i.e., transmitter) application.

Using the “testMP3Receiver.cpp” code as a guide, you would:
	- line 64: Replace this with a call to “AMRAudioFileSink::createNew()"
	- line 70: Replace this string with your stream’s (destination) multicast address, or “0.0.0.0” if it is unicast
	- line 75: Replace this with the incoming stream’s (destination) port number
	- line 107: Replace this with a call to “AMRAudioRTPSource::createNew()”, with relevant parameters.  (In particular, you will want the “isWideband” parameter set to True)

This should be enough to give you a ‘receiver’ application that receives the incoming AMR-WB stream into a file (that you can play with a media player like VLC; you may need to give the filename a “.amr” suffix).

For the transmitter application, you could similarly use “testMP3Streamer” as a guide.  The most important thing to do here is call
	AMRAudioRTPSink::createNew()
instead of
	MPEG1or2AudioRTPSink::createNew()


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