[Live-devel] RTP Audio without sip

Guillermo Bernaldo de Quiros Maraver gbernaldo at cestel.es
Thu Jun 13 06:24:14 PDT 2024


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?

Thank you so much!

El jue, 13 jun 2024 a las 14:42, Ross Finlayson (<finlayson at live555.com>)
escribió:

>
>
> > 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/
>
>
> _______________________________________________
> live-devel mailing list
> live-devel at lists.live555.com
> http://lists.live555.com/mailman/listinfo/live-devel
>


-- 
Guillermo Bernaldo de Quiros Maraver
C/ Marie Curie, 5-7. Edificio
<https://maps.google.com/?q=C/+Marie+Curie,+5-7.+Edificio&entry=gmail&source=g>
Beta,
Oficina 2.5
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Email: gbernaldo at cestel.es
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