[Freeswitch-users] mod_shout delay in trunk
Tamas
jalsot at gmail.com
Fri Feb 6 13:36:51 PST 2009
Hello,
could this option be used to lower I/O load - to rather write more bytes
at once rather than one by one - on file recording (record_session)?
Regards,
Tamas
Anthony Minessale írta:
> Thanks,
>
> We appreciate the positive feedback!
>
> if you revert the change I suggested and update i added a new variable
>
> enable_file_write_buffering=false
>
> set this variable on the channel before you start recording it with
> the set application or in the dialstring in {}
> on outbound calls and it should skip the buffering.
>
> Could you test it for me and confirm it works?
>
> Thank you
>
>
> On Fri, Feb 6, 2009 at 2:36 PM, <freeswitch-users at digitaldan.com
> <mailto:freeswitch-users at digitaldan.com>> wrote:
>
> That worked great!
>
> I wanted to say just how awesome Freeswitch is, I have been doing
> voip related development with SIP since 2000 and this is by far
> the most well designed piece of voip software I have used or
> developed on. I currently have a homegrown sip server built on
> the NIST sip stack with Sun's JMF libraries for RTP processing.
> 95% of the code and complexity is handling the SIP and RTP
> sessions, the other 5% is the final application logic and what is
> most important to me. By letting freeswitch do whats its good at
> (call routing, sip and media handling) it allows me to focus on
> what I'm good at (what should we do with those streams, like
> record them). I have been bragging about this project to anybody
> who will listen!
>
> Dan-
>
> ----- Original Message -----
> From: "Anthony Minessale" <anthony.minessale at gmail.com
> <mailto:anthony.minessale at gmail.com>>
> To: freeswitch-users at lists.freeswitch.org
> <mailto:freeswitch-users at lists.freeswitch.org>
> Sent: Friday, February 6, 2009 1:07:44 PM GMT -07:00 US/Canada
> Mountain
> Subject: Re: [Freeswitch-users] mod_shout delay in trunk
>
> edit switch_ivr_play_say.c line 423
>
> comment the line out and recompile.
> Tell me if it helps you and i will consider making it configurable.
>
>
> On Fri, Feb 6, 2009 at 2:01 PM, <freeswitch-users at digitaldan.com
> <mailto:freeswitch-users at digitaldan.com>> wrote:
>
> For me it is. For what I'm using it for I can tolerate around
> a second or two delay. I have the icecast server setup to
> only buffer 1K for their on-connect burst as well as my
> flash/flex player to only buffer 1k (yes I might as well not
> buffer at all, which I may end up doing). In 1.0.2 this
> worked very well. Is this buffer configurable? If not, where
> is it being set?
>
> Thanks
> Dan-
>
> ----- Original Message -----
> From: "Brian West" <brian at freeswitch.org
> <mailto:brian at freeswitch.org>>
> To: freeswitch-users at lists.freeswitch.org
> <mailto:freeswitch-users at lists.freeswitch.org>
> Sent: Friday, February 6, 2009 12:47:53 PM GMT -07:00
> US/Canada Mountain
> Subject: Re: [Freeswitch-users] mod_shout delay in trunk
>
> Let me clarify.. yes this is normal file buffering was added
> so we wouldn't thrash your hard drive with tiny bits of data
> when recording calls so now it buffers and writes larger
> chunks to disk. This is why you have this delay which is 100%
> normal.... is realtime a critical thing? It is shout cast so
> you know it doesn't have to be realtime.. in fact some clients
> will buffer a little bit anyway and add to it.
>
> /b
>
> On Feb 6, 2009, at 1:43 PM, freeswitch-users at digitaldan.com
> <mailto:freeswitch-users at digitaldan.com> wrote:
>
> I have, do you know what would have changed between 1.0.2
> and trunk that would cause the buffer to change? Also if
> its not in mod_shout.c (which I copied from 1.0.2 to trunk
> for testing with no luck), where else would fs be
> buffering? One thing I have noticed is that in 1.0.2 as
> soon as the dial plan hits my record statement I see
> mod_shout logging that it has connected to the icecast
> server, in trunk it takes about 5 seconds to see the same
> log mesage. Below is my current svn info
> Path: .
>
>
>
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>
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>
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