Re: [Freeswitch-users] mod_shout delay in trunk
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@gmail.com> To: freeswitch-users@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@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@freeswitch.org > To: freeswitch-users@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@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@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@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@hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@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@gmail.com> To: freeswitch-users@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@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@freeswitch.org> To: freeswitch-users@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@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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@gmail.com> To: freeswitch-users@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@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@gmail.com > To: freeswitch-users@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@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@freeswitch.org > To: freeswitch-users@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@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@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@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@hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@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@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@hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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
Also, as payment for services rendered could you please add this variable and description to the wiki? http://wiki.freeswitch.org/wiki/Channel_Variables Thanks! -MC
Done! ----- Original Message ----- From: "Michael Collins" <msc@freeswitch.org> To: freeswitch-users@lists.freeswitch.org Sent: Friday, February 6, 2009 2:36:39 PM GMT -07:00 US/Canada Mountain Subject: Re: [Freeswitch-users] mod_shout delay in trunk
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
Also, as payment for services rendered could you please add this variable and description to the wiki? http://wiki.freeswitch.org/wiki/Channel_Variables Thanks! -MC _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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@digitaldan.com <mailto:freeswitch-users@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@gmail.com <mailto:anthony.minessale@gmail.com>> To: freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@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@digitaldan.com <mailto:freeswitch-users@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@freeswitch.org <mailto:brian@freeswitch.org>> To: freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@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@digitaldan.com <mailto:freeswitch-users@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@lists.freeswitch.org <mailto:Freeswitch-users@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@lists.freeswitch.org <mailto:Freeswitch-users@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@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
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_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400 ------------------------------------------------------------------------
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tamas, the opposite. The default is to not do one by one and setting the var to false makes it more i/o intensive but it would provide more real time recording when recording to streams. BTW the reversed logic is fixed in tree On Fri, Feb 6, 2009 at 3:36 PM, Tamas <jalsot@gmail.com> wrote:
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 II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (5)
-
Anthony Minessale
-
Dan
-
freeswitch-users＠digitaldan.com
-
Michael Collins
-
Tamas