Did you do any debugging info on the other side? What client is it? I can speak from experience that I use the conference with like 10 people in it at a minimum of 8 hours a day and there are never any problems. Try calling your endpoints into the FS public conference to compare sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> this is just a lowly VM On Thu, Mar 25, 2010 at 2:50 PM, Chris Fowler <chris@fowler.cc> wrote:

Forgot to mention : Running trunk – build 17104

*From:* Chris Fowler *Sent:* Thursday, March 25, 2010 12:48 PM *To:* 'freeswitch-users@lists.freeswitch.org' *Subject:* Guidance on debugging audio glitches

Hi,

Here’s the problem I’m “hearing”:

Test Case #1

x1001 calls x1002

- Audio is rock solid

Test Case #2

x1001 calls x3001 (Conference Bridge)

x1002 calls x3001 (Conference Bridge)

- Audio is mostly perfect. However, occasional “glitches” in the audio can be heard

Notes:

- Bypass media is switched off. FreeSWITCH routes audio for both the bridged, and the conferenced call

- rtp-autoflush-during-bridge is “true” on Sofia profile internal.xml

- I enabled the debug trace for auto-flush in switch_rtp.c (line 1895) and see occasional auto flush of rtp packets for **both** test cases. The glitches don’t coincide with the flushing. In fact with flushing you can’t hear when data gets flushed (2x20ms packets).

Environment

- CentOS 5.4

- Tested on AWS EC2 Small, Medium, Large instances. On Medium and Large CPU affinity for FreeSWITCH set to the 2nd core.

Questions:

- What’s different between bridging a call and a conference call run through mod_conference in terms of audio processing?

- Thoughts on how to debug this?

Thanks, Chris.

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