Initiating application jitterbuffer multiple times
Hello, is it OK to start application jitterbuffer multiple times? For example, before bridging a call I would do this: <action application="jitterbuffer" data="60:200:20"/> then if later the call is transferred back to the dialplan, I would call the above again before doing a new bridge. Is there any chance this to be allocating extra resources? Regards, takeshi
When you execute the jitterbuffer application, it will start the jitter buffer. The jitter buffer is paused during bridge by default and is resumed when bridge ends. You can set sip_jitter_buffer_during_bridge=true if you want the jitter buffer to keep running during bridge. It's ok to call the jitterbuffer application multiple times, but what you described didn't make sense to me. When do you want the jitter buffer on? When do you want it paused? Chris On Mon, Aug 20, 2012 at 8:53 PM, mayamatakeshi <mayamatakeshi@gmail.com>wrote:
Hello, is it OK to start application jitterbuffer multiple times? For example, before bridging a call I would do this:
<action application="jitterbuffer" data="60:200:20"/>
then if later the call is transferred back to the dialplan, I would call the above again before doing a new bridge. Is there any chance this to be allocating extra resources?
Regards, takeshi
On Tue, Aug 21, 2012 at 11:24 AM, Christopher Rienzo <cmrienzo@gmail.com>wrote:
When you execute the jitterbuffer application, it will start the jitter buffer. The jitter buffer is paused during bridge by default and is resumed when bridge ends.
OK. Thanks. But this is a bit surprising. What would be the rationale for this? I mean, at least in the company I work for, jitterbuffer is used to solve problems during SIP calls. We are not so far trying to solve any problem with unbridged calls (like calls to voicemail) because we don't have them yet.
You can set sip_jitter_buffer_during_bridge=true if you want the jitter buffer to keep running during bridge.
OK. Noted.
It's ok to call the jitterbuffer application multiple times,
OK.
but what you described didn't make sense to me.
Well, my doubt was if there is any problem of calling the jitterbuffer app multiple times. This will happen in case the channel is sent back to the dialplan (blind transfer) and if I don't check if the jitterbuffer was already set or not. I asked this because I saw things in the past like calling start_dtmf more than once causing duplication of DTMF notification (meaning more than one resource was allocated for the task). So I would just want to confirm this will not be the case for jitterbuffer. Otherwise, I would have to check if jitterbuffer was already called and avoid calling it again (nothing complex, but I will not add extra code if there is not need). (I would set it in the sip profile, but as the wiki says, there you cannot set the jitter buffer max length)
When do you want the jitter buffer on?
Actually I want it enabled all the time no matter if the channel is bridged or not.
When do you want it paused?
Never. Regards, takeshi
On Mon, Aug 20, 2012 at 8:53 PM, mayamatakeshi <mayamatakeshi@gmail.com>wrote:
Hello, is it OK to start application jitterbuffer multiple times? For example, before bridging a call I would do this:
<action application="jitterbuffer" data="60:200:20"/>
then if later the call is transferred back to the dialplan, I would call the above again before doing a new bridge. Is there any chance this to be allocating extra resources?
Regards, takeshi
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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 the jitterbuffer is already on, calling it again will just resize it, so setting it to the same is redundant but harmless. If you are surprised by why the jitterbuffer is paused during bridge: If both sides of a bridge are RTP and both sides have a jb, its fairly useless. In fact if anything, it can worsen call quality. You should only run jitterbuffers at points of termination change of protocol. Examples, if FS was hosting a conference or IVR, if you are bridging the call to a phone for instance, you want to not use a jitterbuffer because you want to preserve the original timestamps so your phone can use its own jitterbuffer. For special examples where you are using FS jitterbuffer in front of something else that may not have one or some other special circumstance you can use the setting chris mentioned to leave it running. On Mon, Aug 20, 2012 at 9:52 PM, mayamatakeshi <mayamatakeshi@gmail.com> wrote:
On Tue, Aug 21, 2012 at 11:24 AM, Christopher Rienzo <cmrienzo@gmail.com> wrote:
When you execute the jitterbuffer application, it will start the jitter buffer. The jitter buffer is paused during bridge by default and is resumed when bridge ends.
OK. Thanks. But this is a bit surprising. What would be the rationale for this? I mean, at least in the company I work for, jitterbuffer is used to solve problems during SIP calls. We are not so far trying to solve any problem with unbridged calls (like calls to voicemail) because we don't have them yet.
You can set sip_jitter_buffer_during_bridge=true if you want the jitter buffer to keep running during bridge.
OK. Noted.
It's ok to call the jitterbuffer application multiple times,
OK.
but what you described didn't make sense to me.
Well, my doubt was if there is any problem of calling the jitterbuffer app multiple times. This will happen in case the channel is sent back to the dialplan (blind transfer) and if I don't check if the jitterbuffer was already set or not. I asked this because I saw things in the past like calling start_dtmf more than once causing duplication of DTMF notification (meaning more than one resource was allocated for the task). So I would just want to confirm this will not be the case for jitterbuffer. Otherwise, I would have to check if jitterbuffer was already called and avoid calling it again (nothing complex, but I will not add extra code if there is not need). (I would set it in the sip profile, but as the wiki says, there you cannot set the jitter buffer max length)
When do you want the jitter buffer on?
Actually I want it enabled all the time no matter if the channel is bridged or not.
When do you want it paused?
Never.
Regards, takeshi
On Mon, Aug 20, 2012 at 8:53 PM, mayamatakeshi <mayamatakeshi@gmail.com> wrote:
Hello, is it OK to start application jitterbuffer multiple times? For example, before bridging a call I would do this:
<action application="jitterbuffer" data="60:200:20"/>
then if later the call is transferred back to the dialplan, I would call the above again before doing a new bridge. Is there any chance this to be allocating extra resources?
Regards, takeshi
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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/ Twitter: http://twitter.com/FreeSWITCH_wire 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 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
On Tue, Aug 21, 2012 at 2:25 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
If the jitterbuffer is already on, calling it again will just resize it, so setting it to the same is redundant but harmless.
If you are surprised by why the jitterbuffer is paused during bridge:
If both sides of a bridge are RTP and both sides have a jb, its fairly useless. In fact if anything, it can worsen call quality.
You should only run jitterbuffers at points of termination change of protocol. Examples, if FS was hosting a conference or IVR, if you are bridging the call to a phone for instance, you want to not use a jitterbuffer because you want to preserve the original timestamps so your phone can use its own jitterbuffer.
Ah. OK. But I failed to mention that I am using FS for transcoding. Our rationale was this: - we have terminals with bad jitter and they can select from a variety of codecs. - FS needs jitter buffers in both sides of the bridge to properly recompose the audio and do the transcoding. Is this OK? If yes, I understand things will not be that simple because then I need to check if transcoding is needed or not and allow for jitterbuffers to only be enabled in this case. Regards, takeshi
For special examples where you are using FS jitterbuffer in front of something else that may not have one or some other special circumstance you can use the setting chris mentioned to leave it running.
On Mon, Aug 20, 2012 at 9:52 PM, mayamatakeshi <mayamatakeshi@gmail.com> wrote:
On Tue, Aug 21, 2012 at 11:24 AM, Christopher Rienzo <cmrienzo@gmail.com
wrote:
When you execute the jitterbuffer application, it will start the jitter buffer. The jitter buffer is paused during bridge by default and is
resumed
when bridge ends.
OK. Thanks. But this is a bit surprising. What would be the rationale for this? I mean, at least in the company I work for, jitterbuffer is used to solve problems during SIP calls. We are not so far trying to solve any problem with unbridged calls (like calls to voicemail) because we don't have them yet.
You can set sip_jitter_buffer_during_bridge=true if you want the jitter buffer to keep running during bridge.
OK. Noted.
It's ok to call the jitterbuffer application multiple times,
OK.
but what you described didn't make sense to me.
Well, my doubt was if there is any problem of calling the jitterbuffer app multiple times. This will happen in case the channel is sent back to the dialplan (blind transfer) and if I don't check if the jitterbuffer was already set or not. I asked this because I saw things in the past like calling start_dtmf more than once causing duplication of DTMF notification (meaning more than one resource was allocated for the task). So I would just want to confirm this will not be the case for jitterbuffer. Otherwise, I would have to check if jitterbuffer was already called and avoid calling it again (nothing complex, but I will not add extra code if there is not need). (I would set it in the sip profile, but as the wiki says, there you cannot set the jitter buffer max length)
When do you want the jitter buffer on?
Actually I want it enabled all the time no matter if the channel is bridged or not.
When do you want it paused?
Never.
Regards, takeshi
On Mon, Aug 20, 2012 at 8:53 PM, mayamatakeshi <mayamatakeshi@gmail.com
wrote:
Hello, is it OK to start application jitterbuffer multiple times? For example, before bridging a call I would do this:
<action application="jitterbuffer" data="60:200:20"/>
then if later the call is transferred back to the dialplan, I would
call
the above again before doing a new bridge. Is there any chance this to be allocating extra resources?
Regards, takeshi
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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/ Twitter: http://twitter.com/FreeSWITCH_wire
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 googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
FYI, this is very useful information so I tossed it up on the wiki on the jitterbuffer page: http://wiki.freeswitch.org/wiki/Jitterbuffer#Interesting_Information If anyone feels like making it look prettier by all means do so. :) -Michael On Mon, Aug 20, 2012 at 10:25 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
If the jitterbuffer is already on, calling it again will just resize it, so setting it to the same is redundant but harmless.
If you are surprised by why the jitterbuffer is paused during bridge:
If both sides of a bridge are RTP and both sides have a jb, its fairly useless. In fact if anything, it can worsen call quality.
You should only run jitterbuffers at points of termination change of protocol. Examples, if FS was hosting a conference or IVR, if you are bridging the call to a phone for instance, you want to not use a jitterbuffer because you want to preserve the original timestamps so your phone can use its own jitterbuffer.
For special examples where you are using FS jitterbuffer in front of something else that may not have one or some other special circumstance you can use the setting chris mentioned to leave it running.
participants (4)
-
Anthony Minessale
-
Christopher Rienzo
-
mayamatakeshi
-
Michael Collins