[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: .
>
>
>
>         _______________________________________________
>         Freeswitch-users mailing list
>         Freeswitch-users at lists.freeswitch.org
>         <mailto:Freeswitch-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>         _______________________________________________
>         Freeswitch-users mailing list
>         Freeswitch-users at lists.freeswitch.org
>         <mailto:Freeswitch-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
>
>     -- 
>     Anthony Minessale II
>
>     FreeSWITCH http://www.freeswitch.org/
>     ClueCon http://www.cluecon.com/
>
>     AIM: anthm
>     MSN:anthony_minessale at hotmail.com
>     <mailto:MSN%3Aanthony_minessale at hotmail.com>
>     GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>     <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>     IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
>     FreeSWITCH Developer Conference
>     sip:888 at conference.freeswitch.org
>     <mailto:sip%3A888 at conference.freeswitch.org>
>     iax:guest at conference.freeswitch.org/888
>     <http://iax:guest@conference.freeswitch.org/888>
>     googletalk:conf+888 at conference.freeswitch.org
>     <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>     pstn:213-799-1400
>
>     _______________________________________________ Freeswitch-users
>     mailing list Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>     _______________________________________________
>     Freeswitch-users mailing list
>     Freeswitch-users at lists.freeswitch.org
>     <mailto:Freeswitch-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest@conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   





More information about the FreeSWITCH-users mailing list