[Freeswitch-users] mod_shout delay in trunk

Fri Feb 6 13:13:06 PST 2009

On line 424 I think it needs to be changed from 

if (!vval || !switch_true(vval)) { 
to 
if (!vval || switch_true(vval)) { 

Other wise it works, thanks! 

----- Original Message ----- 
From: "Anthony Minessale" <anthony.minessale at gmail.com> 
To: freeswitch-users at lists.freeswitch.org 
Sent: Friday, February 6, 2009 1:48:31 PM GMT -07:00 US/Canada Mountain 
Subject: Re: [Freeswitch-users] mod_shout delay in trunk 

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 

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

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-- 
Anthony Minessale II 

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

AIM: anthm 
MSN:anthony_minessale at hotmail.com 
GTALK/JABBER/ PAYPAL:anthony.minessale at gmail.com 
IRC: irc.freenode.net #freeswitch 

FreeSWITCH Developer Conference 
sip:888 at conference.freeswitch.org 
iax:guest at conference.freeswitch.org/888 
googletalk:conf+888 at conference.freeswitch.org 
pstn:213-799-1400 

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