[Live-devel] RTP Audio without sip

[Ross Finlayson]

> On Jun 13, 2024, at 4:44 AM, Guillermo Bernaldo de Quiros Maraver <gbernaldo at cestel.es> wrote:
> 
> The audio comes in RTP format from another place (with their source address, and source port payload type, ssrc, etc...)

[…]

> On the other hand I have to send audio in RTP format when the application sends me a notification. The source of the audio comes from my microphone when I get a notification.
> My question is if there is any possibility to use liveMedia (without SIP) for this kind of scenario. If that's the case, is there any sample code I can use as a guide? 

If I understand correctly, you want to both:
	1/ Receive an incoming RTP audio stream (presumably knowing its port number and destnation/source IP addresses), and
	2/ Transmit an outgoing RTP audio stream (presumably knowing the destination port number and IP address)

If so, you could probably be able to use our “LIVE555 Streaming Media” code for this.  For example, you could use one LIVE555-based application (process) to do 1/ (the reception), and another LIVE555-based application (process) to do 2/ (the transmission).  Or you might want to combine both reception and transmission into a single LIVE555-based application (using a single event loop).

For 1/, do you know what kind of audio (i.e. what codec) this incoming stream is using?  Do you have a SDP description for this stream, or do you always know in advance that the stream will be using one particular codec (and thus one particular RTP payload format), including the sampling frequency, and the number of audio channels (i.e., whether it’s mono or stereo)?  This question is important, because the format of the RTP packet (including possible headers) will depend on the particular codec that is being used.


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