[Freeswitch-users] mod_shout delay in trunk

[Anthony Minessale]

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> 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>
> To: 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> 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>
>> To: 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 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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>
>
> --
> Anthony Minessale II
>
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-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

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