Conference delay increasing over time
I have a very well specified Windows Server connected to sound card input via mod_portaudio. I want multiple callers to be able to listen (only) to the incoming feed, so I set up dynamic conferences, with the dialplan flags setting mute. An autocall bridges the portaudio. My assumption is that this is the best way. I am happy to be wrong if there's a better plan. The problem is that delay builds up, seconds every minute on the conference. I have RTP-Timer-Name=none and enabling both RTP AutoFlushes. Any other suggestions I could try? Would CentOS have better timing? I don't need callers to hear each other, but I'm sure that I'm missing something. Any thoughts? Erik
have you seen mod_portaudio_stream you can use that in a .loc file together with mod_local_stream for static muxing and just play the localstream as a file in your dialplan On Fri, Nov 8, 2013 at 1:26 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I have a very well specified Windows Server connected to sound card input via mod_portaudio.
I want multiple callers to be able to listen (only) to the incoming feed, so I set up dynamic conferences, with the dialplan flags setting mute. An autocall bridges the portaudio.
My assumption is that this is the best way. I am happy to be wrong if there's a better plan.
The problem is that delay builds up, seconds every minute on the conference.
I have RTP-Timer-Name=none and enabling both RTP AutoFlushes.
Any other suggestions I could try? Would CentOS have better timing?
I don't need callers to hear each other, but I'm sure that I'm missing something.
Any thoughts?
Erik
_________________________________________________________________________ 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
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-- 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
No, I hadn't - that sounds good. I'm using the new(ish) PortAudio shstreams endpoints and have been trying to find examples of the .loc approach with multiple soundcards. Any guidance welcomed! Does anyone have any thoughts on why conferences would be slowing down? On Fri, Nov 8, 2013 at 2:00 PM, Anthony Minessale < anthony.minessale@gmail.com <javascript:_e({}, 'cvml', 'anthony.minessale@gmail.com');>> wrote:
have you seen mod_portaudio_stream you can use that in a .loc file together with mod_local_stream for static muxing and just play the localstream as a file in your dialplan
On Fri, Nov 8, 2013 at 1:26 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com <javascript:_e({}, 'cvml', 'emdevane@gmail.com');>>wrote:
I have a very well specified Windows Server connected to sound card input via mod_portaudio.
I want multiple callers to be able to listen (only) to the incoming feed, so I set up dynamic conferences, with the dialplan flags setting mute. An autocall bridges the portaudio.
My assumption is that this is the best way. I am happy to be wrong if there's a better plan.
The problem is that delay builds up, seconds every minute on the conference.
I have RTP-Timer-Name=none and enabling both RTP AutoFlushes.
Any other suggestions I could try? Would CentOS have better timing?
I don't need callers to hear each other, but I'm sure that I'm missing something.
Any thoughts?
Erik
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e({}, 'cvml', '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
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FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <javascript:_e({}, 'cvml', 'MSN%3Aanthony_minessale@hotmail.com');> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <javascript:_e({}, 'cvml', 'PAYPAL%3Aanthony.minessale@gmail.com');> IRC: irc.freenode.net #freeswitch
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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e({}, 'cvml', '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 <javascript:_e({}, 'cvml', '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
Are you running the pa and the conference both at a high rate? Some soundcards do bad at slower rates since its emulated. Its most likely the timing on the soundcard over anything else. On Sat, Nov 9, 2013 at 4:18 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
No, I hadn't - that sounds good. I'm using the new(ish) PortAudio shstreams endpoints and have been trying to find examples of the .loc approach with multiple soundcards.
Any guidance welcomed!
Does anyone have any thoughts on why conferences would be slowing down?
On Fri, Nov 8, 2013 at 2:00 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
have you seen mod_portaudio_stream you can use that in a .loc file together with mod_local_stream for static muxing and just play the localstream as a file in your dialplan
On Fri, Nov 8, 2013 at 1:26 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I have a very well specified Windows Server connected to sound card input via mod_portaudio.
I want multiple callers to be able to listen (only) to the incoming feed, so I set up dynamic conferences, with the dialplan flags setting mute. An autocall bridges the portaudio.
My assumption is that this is the best way. I am happy to be wrong if there's a better plan.
The problem is that delay builds up, seconds every minute on the conference.
I have RTP-Timer-Name=none and enabling both RTP AutoFlushes.
Any other suggestions I could try? Would CentOS have better timing?
I don't need callers to hear each other, but I'm sure that I'm missing something.
Any thoughts?
Erik
_________________________________________________________________________ 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
_________________________________________________________________________ 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
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-- 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
OK, so I've had another chance to work with the settings following your suggestion. Setup - mixing console sending identical channels to sixteen channels on two M-Audio Delta 1010 cards. SIP trunk from Cisco system, testing using AT&T phone to public phone number. CODEC is PCMU/8000. Everything seems perfect when just doing a simple bridge: <action application="bridge" data="portaudio/endpoint/MAUDIO-${destination_number:-2}"/> I can listen to it for hours, with no issues or delay. Problems start when creating a default conference used, by dialplan: <action application="conference_set_auto_outcall" data="portaudio/endpoint/MAUDIO-11"/> <action application="conference" data="$1-${domain_name}@ ${use_profile}++flags{mute}""/> Test - call both bridged extension and conference, and listen. After a minute, there is definite delay in the conference. After three minutes, there is a second delay. After ten minutes, the audio is so far behind it is unusable. Setup: Conference - default, energy level: 100, waste, all callers set to mute. External.xml used for profile, changes to the default: <param name="rtp-timer-name" value="none"/> <param name="rtp-autoflush-during-bridge" value="true"/> <param name="rtp-autoflush" value="true"/> Setting PortAudio rate to 8000, using default conference rate 8000, audio sounds the same, and no issues over bridge. Conference still has delay.. Setting PortAudio rate to 48000, upping default conference rate to 48000, audio sounds the same, and no issues over bridge. Conference still has delay. Any other suggestions would be gladly received. I couldn't locate many examples of using soundcards as an MOH loc and streaming that, so that is another avenue to try, if anyone has any hints. Thank you for your earlier suggestions - that I can stream sixteen channels to the outside world reliably, with sensible configuration options, is an outstanding achievement by the developers. Now if I could have multiple callers receive the same audio... On Sat, Nov 9, 2013 at 7:20 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Are you running the pa and the conference both at a high rate? Some soundcards do bad at slower rates since its emulated. Its most likely the timing on the soundcard over anything else.
On Sat, Nov 9, 2013 at 4:18 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
No, I hadn't - that sounds good. I'm using the new(ish) PortAudio shstreams endpoints and have been trying to find examples of the .loc approach with multiple soundcards.
Any guidance welcomed!
Does anyone have any thoughts on why conferences would be slowing down?
On Fri, Nov 8, 2013 at 2:00 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
have you seen mod_portaudio_stream you can use that in a .loc file together with mod_local_stream for static muxing and just play the localstream as a file in your dialplan
On Fri, Nov 8, 2013 at 1:26 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I have a very well specified Windows Server connected to sound card input via mod_portaudio.
I want multiple callers to be able to listen (only) to the incoming feed, so I set up dynamic conferences, with the dialplan flags setting mute. An autocall bridges the portaudio.
My assumption is that this is the best way. I am happy to be wrong if there's a better plan.
The problem is that delay builds up, seconds every minute on the conference.
I have RTP-Timer-Name=none and enabling both RTP AutoFlushes.
Any other suggestions I could try? Would CentOS have better timing?
I don't need callers to hear each other, but I'm sure that I'm missing something.
Any thoughts?
Erik
_________________________________________________________________________ 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
_________________________________________________________________________ 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
I've been working at this all day, with no joy. I thought that antivirus software was to blame, but that was a dead end. Does anyone have an example of running the .loc local_stream approach? On Sunday, November 10, 2013, Erik M. Devane - Comms Guy wrote:
OK, so I've had another chance to work with the settings following your suggestion.
Setup - mixing console sending identical channels to sixteen channels on two M-Audio Delta 1010 cards. SIP trunk from Cisco system, testing using AT&T phone to public phone number. CODEC is PCMU/8000.
Everything seems perfect when just doing a simple bridge: <action application="bridge" data="portaudio/endpoint/MAUDIO-${destination_number:-2}"/> I can listen to it for hours, with no issues or delay.
Problems start when creating a default conference used, by dialplan:
<action application="conference_set_auto_outcall" data="portaudio/endpoint/MAUDIO-11"/> <action application="conference" data="$1-${domain_name}@ ${use_profile}++flags{mute}""/>
Test - call both bridged extension and conference, and listen. After a minute, there is definite delay in the conference. After three minutes, there is a second delay. After ten minutes, the audio is so far behind it is unusable.
Setup:
Conference - default, energy level: 100, waste, all callers set to mute.
External.xml used for profile, changes to the default:
<param name="rtp-timer-name" value="none"/> <param name="rtp-autoflush-during-bridge" value="true"/> <param name="rtp-autoflush" value="true"/>
Setting PortAudio rate to 8000, using default conference rate 8000, audio sounds the same, and no issues over bridge. Conference still has delay..
Setting PortAudio rate to 48000, upping default conference rate to 48000, audio sounds the same, and no issues over bridge. Conference still has delay.
Any other suggestions would be gladly received. I couldn't locate many examples of using soundcards as an MOH loc and streaming that, so that is another avenue to try, if anyone has any hints.
Thank you for your earlier suggestions - that I can stream sixteen channels to the outside world reliably, with sensible configuration options, is an outstanding achievement by the developers. Now if I could have multiple callers receive the same audio...
On Sat, Nov 9, 2013 at 7:20 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Are you running the pa and the conference both at a high rate? Some soundcards do bad at slower rates since its emulated. Its most likely the timing on the soundcard over anything else.
On Sat, Nov 9, 2013 at 4:18 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
No, I hadn't - that sounds good. I'm using the new(ish) PortAudio shstreams endpoints and have been trying to find examples of the .loc approach with multiple soundcards.
Any guidance welcomed!
Does anyone have any thoughts on why conferences would be slowing down?
On Fri, Nov 8, 2013 at 2:00 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
have you seen mod_portaudio_stream you can use that in a .loc file together with mod_local_stream for static muxing and just play the localstream as a file in your dialplan
On Fri, Nov 8, 2013 at 1:26 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I have a very well specified Windows Server connected to sound card input via mod_portaudio.
I want multiple callers to be able to listen (only) to the incoming feed, so I set up dynamic conferences, with the dialplan flags setting mute. An autocall bridges the portaudio.
My assumption is that this is the best way. I am happy to be wrong if there's a better plan.
The problem is that delay builds up, seconds every minute on the conference.
I have RTP-Timer-Name=none and enabling both RTP AutoFlushes.
Any other suggestions I could try? Would CentOS have better timing?
I don't need callers to hear each other, but I'm sure that I'm missing something.
Any thoughts?
Erik
_________________________________________________________________________ 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
https://wiki.freeswitch.org/wiki/Mod_local_stream#moh.loc Its just setting up mod_local_stream to point at a dir with a file with a .loc extension that has the string of the pa url in it. You should probably not change any of those params away from default. I would recommend putting those back to where they belong. If anything it could worsen your problem. What about the interval? DId you try increasing that? On Mon, Nov 11, 2013 at 11:00 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I've been working at this all day, with no joy.
I thought that antivirus software was to blame, but that was a dead end.
Does anyone have an example of running the .loc local_stream approach?
On Sunday, November 10, 2013, Erik M. Devane - Comms Guy wrote:
OK, so I've had another chance to work with the settings following your suggestion.
Setup - mixing console sending identical channels to sixteen channels on two M-Audio Delta 1010 cards. SIP trunk from Cisco system, testing using AT&T phone to public phone number. CODEC is PCMU/8000.
Everything seems perfect when just doing a simple bridge: <action application="bridge" data="portaudio/endpoint/MAUDIO-${destination_number:-2}"/> I can listen to it for hours, with no issues or delay.
Problems start when creating a default conference used, by dialplan:
<action application="conference_set_auto_outcall" data="portaudio/endpoint/MAUDIO-11"/> <action application="conference" data="$1-${domain_name}@ ${use_profile}++flags{mute}""/>
Test - call both bridged extension and conference, and listen. After a minute, there is definite delay in the conference. After three minutes, there is a second delay. After ten minutes, the audio is so far behind it is unusable.
Setup:
Conference - default, energy level: 100, waste, all callers set to mute.
External.xml used for profile, changes to the default:
<param name="rtp-timer-name" value="none"/> <param name="rtp-autoflush-during-bridge" value="true"/> <param name="rtp-autoflush" value="true"/>
Setting PortAudio rate to 8000, using default conference rate 8000, audio sounds the same, and no issues over bridge. Conference still has delay..
Setting PortAudio rate to 48000, upping default conference rate to 48000, audio sounds the same, and no issues over bridge. Conference still has delay.
Any other suggestions would be gladly received. I couldn't locate many examples of using soundcards as an MOH loc and streaming that, so that is another avenue to try, if anyone has any hints.
Thank you for your earlier suggestions - that I can stream sixteen channels to the outside world reliably, with sensible configuration options, is an outstanding achievement by the developers. Now if I could have multiple callers receive the same audio...
On Sat, Nov 9, 2013 at 7:20 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Are you running the pa and the conference both at a high rate? Some soundcards do bad at slower rates since its emulated. Its most likely the timing on the soundcard over anything else.
On Sat, Nov 9, 2013 at 4:18 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
No, I hadn't - that sounds good. I'm using the new(ish) PortAudio shstreams endpoints and have been trying to find examples of the .loc approach with multiple soundcards.
Any guidance welcomed!
Does anyone have any thoughts on why conferences would be slowing down?
On Fri, Nov 8, 2013 at 2:00 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
have you seen mod_portaudio_stream you can use that in a .loc file together with mod_local_stream for static muxing and just play the localstream as a file in your dialplan
On Fri, Nov 8, 2013 at 1:26 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I have a very well specified Windows Server connected to sound card input via mod_portaudio.
I want multiple callers to be able to listen (only) to the incoming feed, so I set up dynamic conferences, with the dialplan flags setting mute. An autocall bridges the portaudio.
My assumption is that this is the best way. I am happy to be wrong if there's a better plan.
The problem is that delay builds up, seconds every minute on the conference.
I have RTP-Timer-Name=none and enabling both RTP AutoFlushes.
Any other suggestions I could try? Would CentOS have better timing?
I don't need callers to hear each other, but I'm sure that I'm missing something.
Any thoughts?
Erik
_________________________________________________________________________ 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
_________________________________________________________________________ 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
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-- 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
Increasing the conference internal to 10 to 20 to 200 had no detectable effect. In order to improve timing, I rebuilt the server as Centos 6.4, building from the tarball (git had some proxy-related issues). [New issue on Linux] I haven't been able to get mod_portaudio to give me more than two mono inputs per M-Audio Delta 1010 card (though alsamixer and pa_devs seem perfectly happy). [Back to the original issue] I reverted all RTP timer, flush, and other changes, back very close to the default configuration. Still I have a delay, still increases seconds per minute. I explored moh but couldn't figure out how to have multiple playlists. If this is possible, Id love to know how! My best solution so far has been to have the first call bridge portaudio and subsequent calls eavesdrop on that first bridged call. Latency is almost as good as a single bridged call, and I have tested six eavesdroppers with no ill effects. Right now I'm going to: 1) Work out how to mute external caller's audio from the eavesdrop. 2) Learn lua enough to roll an eavesdropper onto the bridge when the first caller hangs up. It's not a satisfactory solution, but I'm out of ideas for the conference lag unless anyone else can help, and I don't know how to do multiple MOH instances. Any help would be most appreciated - either on conference lag or moh or eavesdrop mute. Erik On Tuesday, November 12, 2013, Anthony Minessale wrote:
https://wiki.freeswitch.org/wiki/Mod_local_stream#moh.loc
Its just setting up mod_local_stream to point at a dir with a file with a .loc extension that has the string of the pa url in it.
You should probably not change any of those params away from default. I would recommend putting those back to where they belong.
If anything it could worsen your problem. What about the interval? DId you try increasing that?
On Mon, Nov 11, 2013 at 11:00 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I've been working at this all day, with no joy.
I thought that antivirus software was to blame, but that was a dead end.
Does anyone have an example of running the .loc local_stream approach?
On Sunday, November 10, 2013, Erik M. Devane - Comms Guy wrote:
OK, so I've had another chance to work with the settings following your suggestion.
Setup - mixing console sending identical channels to sixteen channels on two M-Audio Delta 1010 cards. SIP trunk from Cisco system, testing using AT&T phone to public phone number. CODEC is PCMU/8000.
Everything seems perfect when just doing a simple bridge: <action application="bridge" data="portaudio/endpoint/MAUDIO-${destination_number:-2}"/> I can listen to it for hours, with no issues or delay.
Problems start when creating a default conference used, by dialplan:
<action application="conference_set_auto_outcall" data="portaudio/endpoint/MAUDIO-11"/> <action application="conference" data="$1-${domain_name}@ ${use_profile}++flags{mute}""/>
Test - call both bridged extension and conference, and listen. After a minute, there is definite delay in the conference. After three minutes, there is a second delay. After ten minutes, the audio is so far behind it is unusable.
Setup:
Conference - default, energy level: 100, waste, all callers set to mute.
External.xml used for profile, changes to the default:
<param name="rtp-timer-name" value="none"/> <param name="rtp-autoflush-during-bridge" value="true"/> <param name="rtp-autoflush" value="true"/>
Setting PortAudio rate to 8000, using default conference rate 8000, audio sounds the same, and no issues over bridge. Conference still has delay..
Setting PortAudio rate to 48000, upping default conference rate to 48000, audio sounds the same, and no issues over bridge. Conference still has delay.
Any other suggestions would be gladly received. I couldn't locate many examples of using soundcards as an MOH loc and streaming that, so that is another avenue to try, if anyone has any hints.
Thank you for your earlier suggestions - that I can stream sixteen channels to the outside world reliably, with sensible configuration options, is an outstanding achievement by the developers. Now if I could have multiple callers receive the same audio...
On Sat, Nov 9, 2013 at 7:20 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Are you running the pa and the conference both at a high rate? Some soundcards do bad at slower rates since its emulated. Its most likely the timing on the soundcard over anything else.
On Sat, Nov 9, 2013 at 4:18 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
No, I hadn't - that sounds good. I'm using the new(ish) PortAudio shstreams endpoints and have been trying to find examples of the .loc approach with multiple soundcards.
Any guidance welcomed!
Does anyone have any thoughts on why conferences would be slowing down?
On Fri, Nov 8, 2013 at 2:00 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
have you seen mod_portaudio_stream you can use that in a .loc file together with mod_local_stream for static muxing and just play the localstream as a file in your dialplan
On Fri, Nov 8, 2013 at 1:26 PM, Erik M. Devane - Comms Guy <emdevane@g
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <javascript:_e({}, 'cvml', 'sip%3A888@conference.freeswitch.org');> googletalk:conf+888@conference.freeswitch.org <javascript:_e({}, 'cvml', 'googletalk%3Aconf%2B888@conference.freeswitch.org');> pstn:+19193869900
I am not seeing this here at all using real hardware and latest git as of a few days ago. I was getting a year maybe or so ago (not as bad as yours), but it went away and I never figured out why. Erik M. Devane - Comms Guy <emdevane@gmail.com> wrote:
Increasing the conference internal to 10 to 20 to 200 had no detectable effect.
In order to improve timing, I rebuilt the server as Centos 6.4, building from the tarball (git had some proxy-related issues).
[New issue on Linux] I haven't been able to get mod_portaudio to give me more than two mono inputs per M-Audio Delta 1010 card (though alsamixer and pa_devs seem perfectly happy).
[Back to the original issue] I reverted all RTP timer, flush, and other changes, back very close to the default configuration.
Still I have a delay, still increases seconds per minute.
I explored moh but couldn't figure out how to have multiple playlists. If this is possible, Id love to know how!
My best solution so far has been to have the first call bridge portaudio and subsequent calls eavesdrop on that first bridged call. Latency is almost as good as a single bridged call, and I have tested six eavesdroppers with no ill effects.
Right now I'm going to:
1) Work out how to mute external caller's audio from the eavesdrop.
2) Learn lua enough to roll an eavesdropper onto the bridge when the first caller hangs up.
It's not a satisfactory solution, but I'm out of ideas for the conference lag unless anyone else can help, and I don't know how to do multiple MOH instances.
Any help would be most appreciated - either on conference lag or moh or eavesdrop mute.
Erik
On Tuesday, November 12, 2013, Anthony Minessale wrote:
https://wiki.freeswitch.org/wiki/Mod_local_stream#moh.loc
Its just setting up mod_local_stream to point at a dir with a file with a .loc extension that has the string of the pa url in it.
You should probably not change any of those params away from default. I would recommend putting those back to where they belong.
If anything it could worsen your problem. What about the interval? DId you try increasing that?
On Mon, Nov 11, 2013 at 11:00 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I've been working at this all day, with no joy.
I thought that antivirus software was to blame, but that was a dead end.
Does anyone have an example of running the .loc local_stream approach?
On Sunday, November 10, 2013, Erik M. Devane - Comms Guy wrote:
OK, so I've had another chance to work with the settings following your suggestion.
Setup - mixing console sending identical channels to sixteen channels on two M-Audio Delta 1010 cards. SIP trunk from Cisco system, testing using AT&T phone to public phone number. CODEC is PCMU/8000.
Everything seems perfect when just doing a simple bridge: <action application="bridge" data="portaudio/endpoint/MAUDIO-${destination_number:-2}"/> I can listen to it for hours, with no issues or delay.
Problems start when creating a default conference used, by dialplan:
<action application="conference_set_auto_outcall" data="portaudio/endpoint/MAUDIO-11"/> <action application="conference" data="$1-${domain_name}@ ${use_profile}++flags{mute}""/>
Test - call both bridged extension and conference, and listen. After a minute, there is definite delay in the conference. After three minutes, there is a second delay. After ten minutes, the audio is so far behind it is unusable.
Setup:
Conference - default, energy level: 100, waste, all callers set to mute.
External.xml used for profile, changes to the default:
<param name="rtp-timer-name" value="none"/> <param name="rtp-autoflush-during-bridge" value="true"/> <param name="rtp-autoflush" value="true"/>
Setting PortAudio rate to 8000, using default conference rate 8000, audio sounds the same, and no issues over bridge. Conference still has delay..
Setting PortAudio rate to 48000, upping default conference rate to 48000, audio sounds the same, and no issues over bridge. Conference still has delay.
Any other suggestions would be gladly received. I couldn't locate many examples of using soundcards as an MOH loc and streaming that, so that is another avenue to try, if anyone has any hints.
Thank you for your earlier suggestions - that I can stream sixteen channels to the outside world reliably, with sensible configuration options, is an outstanding achievement by the developers. Now if I could have multiple callers receive the same audio...
On Sat, Nov 9, 2013 at 7:20 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Are you running the pa and the conference both at a high rate? Some soundcards do bad at slower rates since its emulated. Its most likely the timing on the soundcard over anything else.
On Sat, Nov 9, 2013 at 4:18 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
No, I hadn't - that sounds good. I'm using the new(ish) PortAudio shstreams endpoints and have been trying to find examples of the .loc approach with multiple soundcards.
Any guidance welcomed!
Does anyone have any thoughts on why conferences would be slowing down?
On Fri, Nov 8, 2013 at 2:00 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
have you seen mod_portaudio_stream you can use that in a .loc file together with mod_local_stream for static muxing and just play the localstream as a file in your dialplan
On Fri, Nov 8, 2013 at 1:26 PM, Erik M. Devane - Comms Guy <emdevane@g
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <javascript:_e({}, 'cvml', 'sip%3A888@conference.freeswitch.org');> googletalk:conf+888@conference.freeswitch.org <javascript:_e({}, 'cvml', 'googletalk%3Aconf%2B888@conference.freeswitch.org');> pstn:+19193869900
---------------------------------------------------- Alternatives:
---------------------------------------------------- _________________________________________________________________________ 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
-- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com
I am doing a similar setup on a Windows Server using conference with some C# code.In my system I have a few extension each one is playing music from a different web stream. The way I set it up is that when a call comes in I check if the conference related to the dialed extension is already setup ("conference <extension> list" doesn't return "not found"), if not I create a new thread and execute "conference <extension> play <url> async" and then conference in the caller. If the conference is already in session I just add the caller to the conference. The reason I do it on a separate thread is that conference play only works once the conference is already created, but the api call to add the caller to the conference is a blocking call and will only return when the user hangs up or is kicked from the conference. I also had some issues with the stream falling behind, also the stream would sometimes go silent, so I resolved this by setting a time for every 5 minutes which executes "conference <extension> stop" and then "conf <extension> play <url>". This happens so quickly that the listeners barely notice that the stream has been restarted. You might be able to implement a similar setup with other languages. I went through a few iterations until I got to this setup and it now appears to be working great. On Fri, Nov 22, 2013 at 7:55 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
Increasing the conference internal to 10 to 20 to 200 had no detectable effect.
In order to improve timing, I rebuilt the server as Centos 6.4, building from the tarball (git had some proxy-related issues).
[New issue on Linux] I haven't been able to get mod_portaudio to give me more than two mono inputs per M-Audio Delta 1010 card (though alsamixer and pa_devs seem perfectly happy).
[Back to the original issue] I reverted all RTP timer, flush, and other changes, back very close to the default configuration.
Still I have a delay, still increases seconds per minute.
I explored moh but couldn't figure out how to have multiple playlists. If this is possible, Id love to know how!
My best solution so far has been to have the first call bridge portaudio and subsequent calls eavesdrop on that first bridged call. Latency is almost as good as a single bridged call, and I have tested six eavesdroppers with no ill effects.
Right now I'm going to:
1) Work out how to mute external caller's audio from the eavesdrop.
2) Learn lua enough to roll an eavesdropper onto the bridge when the first caller hangs up.
It's not a satisfactory solution, but I'm out of ideas for the conference lag unless anyone else can help, and I don't know how to do multiple MOH instances.
Any help would be most appreciated - either on conference lag or moh or eavesdrop mute.
Erik
On Tuesday, November 12, 2013, Anthony Minessale wrote:
https://wiki.freeswitch.org/wiki/Mod_local_stream#moh.loc
Its just setting up mod_local_stream to point at a dir with a file with a .loc extension that has the string of the pa url in it.
You should probably not change any of those params away from default. I would recommend putting those back to where they belong.
If anything it could worsen your problem. What about the interval? DId you try increasing that?
On Mon, Nov 11, 2013 at 11:00 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I've been working at this all day, with no joy.
I thought that antivirus software was to blame, but that was a dead end.
Does anyone have an example of running the .loc local_stream approach?
On Sunday, November 10, 2013, Erik M. Devane - Comms Guy wrote:
OK, so I've had another chance to work with the settings following your suggestion.
Setup - mixing console sending identical channels to sixteen channels on two M-Audio Delta 1010 cards. SIP trunk from Cisco system, testing using AT&T phone to public phone number. CODEC is PCMU/8000.
Everything seems perfect when just doing a simple bridge: <action application="bridge" data="portaudio/endpoint/MAUDIO-${destination_number:-2}"/> I can listen to it for hours, with no issues or delay.
Problems start when creating a default conference used, by dialplan:
<action application="conference_set_auto_outcall" data="portaudio/endpoint/MAUDIO-11"/> <action application="conference" data="$1-${domain_name}@ ${use_profile}++flags{mute}""/>
Test - call both bridged extension and conference, and listen. After a minute, there is definite delay in the conference. After three minutes, there is a second delay. After ten minutes, the audio is so far behind it is unusable.
Setup:
Conference - default, energy level: 100, waste, all callers set to mute.
External.xml used for profile, changes to the default:
<param name="rtp-timer-name" value="none"/> <param name="rtp-autoflush-during-bridge" value="true"/> <param name="rtp-autoflush" value="true"/>
Setting PortAudio rate to 8000, using default conference rate 8000, audio sounds the same, and no issues over bridge. Conference still has delay..
Setting PortAudio rate to 48000, upping default conference rate to 48000, audio sounds the same, and no issues over bridge. Conference still has delay.
Any other suggestions would be gladly received. I couldn't locate many examples of using soundcards as an MOH loc and streaming that, so that is another avenue to try, if anyone has any hints.
Thank you for your earlier suggestions - that I can stream sixteen channels to the outside world reliably, with sensible configuration options, is an outstanding achievement by the developers. Now if I could have multiple callers receive the same audio...
On Sat, Nov 9, 2013 at 7:20 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
Are you running the pa and the conference both at a high rate? Some soundcards do bad at slower rates since its emulated. Its most likely the timing on the soundcard over anything else.
On Sat, Nov 9, 2013 at 4:18 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
No, I hadn't - that sounds good. I'm using the new(ish) PortAudio shstreams endpoints and have been trying to find examples of the .loc approach with multiple soundcards.
Any guidance welcomed!
Does anyone have any thoughts on why conferences would be slowing down?
On Fri, Nov 8, 2013 at 2:00 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
have you seen mod_portaudio_stream you can use that in a .loc file together with mod_local_stream for static muxing and just play the localstream as a file in your dialplan
On Fri, Nov 8, 2013 at 1:26 PM, Erik M. Devane - Comms Guy <emdevane@g
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
I gave you the wiki page to mod_local_stream quite some time ago, loc files are just ascii files with a list of files to play which is where you put the stream url from pa. If you are looking for no latency that may not be good since it buffers.... On Nov 9, 2013 4:22 PM, "Erik M. Devane - Comms Guy" <emdevane@gmail.com> wrote:
No, I hadn't - that sounds good. I'm using the new(ish) PortAudio shstreams endpoints and have been trying to find examples of the .loc approach with multiple soundcards.
Any guidance welcomed!
Does anyone have any thoughts on why conferences would be slowing down?
On Fri, Nov 8, 2013 at 2:00 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
have you seen mod_portaudio_stream you can use that in a .loc file together with mod_local_stream for static muxing and just play the localstream as a file in your dialplan
On Fri, Nov 8, 2013 at 1:26 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I have a very well specified Windows Server connected to sound card input via mod_portaudio.
I want multiple callers to be able to listen (only) to the incoming feed, so I set up dynamic conferences, with the dialplan flags setting mute. An autocall bridges the portaudio.
My assumption is that this is the best way. I am happy to be wrong if there's a better plan.
The problem is that delay builds up, seconds every minute on the conference.
I have RTP-Timer-Name=none and enabling both RTP AutoFlushes.
Any other suggestions I could try? Would CentOS have better timing?
I don't need callers to hear each other, but I'm sure that I'm missing something.
Any thoughts?
Erik
_________________________________________________________________________ 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
_________________________________________________________________________ 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
participants (4)
-
Anthony Minessale
-
covici＠ccs.covici.com
-
Duvid Rottenberg
-
Erik M. Devane - Comms Guy