On 08/17/2012 04:58 AM, Steve Underwood wrote:
On 08/17/2012 06:08 AM, Eric Wieling wrote:
Has anyone experimented with increasing the DAHDI chunk size in
improve fax reliability? If so, did it help, hurt, or not make any
difference?
I haven't found issues related to the DAHDI chunk size. The main thing
which used to hurt FAXing with Asterisk before Digium launched their
own FAX software was the timing within Asterisk, which they refused to
fix at that time (although independent patches were available). With
the launch of FFA they changed chan_dahdi so on a FAX call the
buffering should change to make the flow of transmitted audio a lot
more elastic. People just tolerate some hiccups in voice calls, but
hate latency. Modem signals must be rigidly timed, but a bit more
latency is OK. This change fixed the main issue affecting all the FAX
solutions around. If that switch in the buffering mode is not
happening on your system for some reason it can badly affect the
reliability of FAXes.
I'm uncertain of exactly to which changes you're referring. Your
comments seem to fall in-line with the notion behind the DAHDI "buffers"
feature for the channel as well as the DAHDI fax-detection "faxbuffers"
feature, but I'm seeing no noticeable improvement, AND I'm uncertain how
to implement the CHANNEL(buffers) feature due to:
-- Executing [4628160@fax-outbound:1] Set("IAX2/ttyIAX99-584",
"CHANNEL(buffers)="12,half"") in new stack
[Aug 18 20:12:40] WARNING[6381]: func_channel.c:530
func_channel_write_real: Unknown or unavailable item requested: 'buffers'
-- Executing [4628160@fax-outbound:2] Goto("IAX2/ttyIAX99-584",
"outbound,4628160,1") in new stack
-- Goto (outbound,4628160,1)
-- Executing [4628160@outbound:1] Dial("IAX2/ttyIAX99-584",
"DAHDI/g0/4628160") in new stack
On some installations there are occasional instances in most outbound
calls where Asterisk creates what otherwise would be considered jitter
on the DAHDI channel. Generally these do not cause much real-world
trouble, but I'm a stickler for perfect audio quality on all-digital
calls. I've seen this on Asterisk versions 1.4, 1.6, and 1.8. On other
installations there never is any such trouble noticeable.
Would you mind being a bit more specific on the Asterisk changes to
which you refer and how they should be implemented in the configuration?
Thanks,
Lee.
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