[Freeswitch-users] one way audio - another report related to auto-jitter-buffer-msec possible behavior change with 1.6

[Nathan Neulinger]

Just wanted to add an additional data point on the one way audio issue (as it turns out - also with grandstreams) with 1.6.

	AnalogDevice <-> Grandstream <-> FS <-> SIP-Provider

One my most recent test/capture/etc, the behavior I saw was that it looked to me like audio stopped being sent from 
freeswitch to outside/external leg once early media finished. Note that the grandstream was continuing to send RTP to 
FS, but FS wasn't passing any of it on to the provider.

Once I saw the thread from the other user having this problem, tried turning it off on my profiles that had it enabled, 
and the problem appears to have gone away. I can provide logs/debug/captures/etc. if requested from a test environment.

I did not have any reports of this issue except related to grandstreams (ht-701) - in my case, almost all related to 
analog faxes (bulk of our ATA usage). Also have not been able to verify the fix yet on a call originating from the 
grandstream (testing from home).

-- Nathan

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Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect