[Asterisk-Dev] Accept patch to make iax2's calc_timestamp suspicious about the ast_frame.delivery value?

[Andrew Kohlsmith]

On Friday 18 June 2004 03:10, steve at daviesfam.org wrote:
> This is bad when you use the jitter buffer as the buffer starts tossing
> away all these "very old" frames and the result is silence...

Could two * boxes over a ~4ms link manage to bugger up the timestamps for 
three or four seconds?  I had an issue with my setup in where connections 
going to the VOIP provider but not through the remote PRI would now and again 
get three seconds of absolute dead silence in one direction only.  After the 
"gap" audio returned to normal.

Path (no NAT, middle * box never got out of the loop since lefthand * box was 
on private LAN with middle * box's 2nd ethernet adapter):

Nortel MICS -> Adit600 -> T100P(*1) -> IAX2 -> (*2) -> IAX2 -> Nufone

*2 also has a TE405P with one port going to a telco PRI for local 
incoming/outgoing calls.  There was never dead audio on calls going in/out 
the PRI, only through Nufone.

Jeremy and friends suggested disabling jitter buffer between *1 and *2, 
switching to GSM (from iLBC) and enabling trunking.  I had disabled trunking 
on *2 entirely since with jitter buffer enabled, trunking turned regular 
audio into what I would call "bursty" audio:

Normal: "Hi, how are you today?"
Bursty:  "Hihow....areyouto....day?"

Nufone's suggestions fixed the problem but I'm curious as to what fixed it.  I 
hope to turn things back to the original way one thing at a time to see what 
fixes it but the dropouts were an infrequent thing and the system seems to 
just work now, so I'm hesitant to bugger around with it on a production 
system.  :-)

-A.