RTP and RTCP are different protocols, but which work together. RTP packets
deliver the media data. RTCP is used (in the reverse direction) to
monitor/report the quality of the RTP stream (including reporting packet loss
rates). Also, if you have separate RTP streams for audio and video, then
Hi Ross,
I've been reading the source code and trying the tests
(testAMRAudioStreamer.cpp as you told me on your last answer). One last
thing I would like to ask you is if there is anyway to use RTP instead of
RTCP. The test I'm using right now, works as expected but I would like to
use RTP becaus
Hi Ross!
Thank you so much, I'm working on the receiver as you told me! I suppose
you will have news (good news I hope) from me as soon as I start testing
the program. Thank you so much!
El jue, 13 jun 2024 a las 17:17, Ross Finlayson ()
escribió:
>
>
> > On Jun 13, 2024, at 7:20 AM, Ross Finlay
> On Jun 13, 2024, at 6:24 AM, Guillermo Bernaldo de Quiros Maraver
> 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 med
> On Jun 13, 2024, at 7:20 AM, Ross Finlayson wrote:
>
> For the transmitter application, you could similarly use “testMP3Streamer” as
> a guide.
Or you could use the “testAMRAudioStreamer” application.
Ross Finlayson
Live Networks, Inc.
http://www.live555.com/
___
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 s
> On Jun 13, 2024, at 4:44 AM, Guillermo Bernaldo de Quiros Maraver
> 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
> send
Hi, good morning!
I have a question regarding using liveMedia without sip. I mean, I have one
application which handles SIP sessions and that application sends me
notifications about RTP Flows (just notifications about starting/stopping a
new RTP Flows, ...) but that application does not handle RT
