On Mon, Feb 23, 2009 at 1:00 PM, Andreas Ahland <a.ahland at gmx.de> wrote: > Hi forks, > > I am using a streaming media approach using the conference bridge which is > streaming RTP streams. When connecting an USB sound device, I see 5 to 6 > 20ms packets being stored and sent on one rush. This adds up unnecessary > delays. Does any have a solution or an idea? Is it possible to define a > master port and poll the sound device? Or is there a more easy way to > configure the sound device to send the data more timely? > > Hi Andreas, so far I haven't found any ways to configure the sound device to send data more timely, I tend to think that this is just the characteristic of that sound device. At least on Windows. I tried to switch between WMME and DirectSound, the burst characteristic is the same. Though you can experiment with giving larger buffer size for the sound device (PJMEDIA_SND_DEFAULT_REC_LATENCY) and see if that improves it. See http://trac.pjsip.org/repos/wiki/FAQ#tx-timing for the rest of your questions. Though I would think that it would not improve the latency, as the master port will do the buffering instead of the jitter buffer in the receiver's end. So it's just moving the delay to the transmitter's end. cheers Benny -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20090224/519261f7/attachment.html>
