[pulseaudio-discuss] Why not RTP instead of TCP

Tanu Kaskinen tanu.kaskinen at linux.intel.com
Wed Aug 28 04:36:34 PDT 2013


On Tue, 2013-08-27 at 10:15 -0400, brian mullan wrote:
> New to this mailer so ignore if this has been beat like an old rug
> already...
> 
> I have been experimenting with xrdp/x11rdp and freerdp in particular how to
> get audio from an rdp remote desktop when that desktop is linux.

Have you talked with Jay Sorg? He was writing PulseAudio support for
xrdp too. In his solution PulseAudio's network protocol is irrelevant,
because the over-the-network part is implemented by the RDP stack, not
PulseAudio.

> Looking at pulseaudio docs it appears it uses only TCP.

There are modules also for RTP streaming, but yes, we mostly use TCP.

> Was there ever any thought to implementing RTP (using UDP) instead?

There has certainly been thought. The native protocol is tied to TCP as
long as the audio streams go over the same stream as the control
messages. It would be awesome if someone would remove this coupling, but
even if that was done, it's not clear that we'd use RTP for the native
streaming.

> Audio is more often one-way and dropping a few UDP audio packets every so
> often would probably not be heard/recognized by average human listener.
> 
> But TCP requiring retransmissions of all dropped packets can introduce
> delays that to me, is often much more noticable.

I find it unlikely that dropped packets wouldn't be easily audible,
unless the codec specifically supports error correction (like Opus, for
example). But yes, TCP probably causes worse effects, if the buffering
is insufficient for dealing with network delays.

-- 
Tanu



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