I've been working on using FreeSWITCH as a bridge between different devices, and I need to delay all calls by 2-3 seconds in order for the external systems to keep up. This delay should happen at the start of a call, and all subsequent audio should sound fine, just late. I know that doing this sounds silly - I'm sorry about that. Here are my approaches: 1) Eavesdrop on a call send through delay_echo - this works for SIP clients but I have to do an absolute_codec_string as it seems that CODEC support is limited, and I'm often getting no delay at all with my PortAudio clients. 2) Use lua to do a session freeswitch.msleep on answer - exactly what Confluence tells me to avoid - if I sleep but don't consume audio, I think I can get the desired result. 3) Mess with the jitterbuffer - I haven't had any success with this yet. 4) Increase the buffer in mod_PortAudio. Writing this out makes me sure that I'm missing something - a way of getting delay_echo to work for all calls, or a way to pause session media to get the desired effect. Can you please help?
JB should be fine. Any details of unsuccess? https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer On Thu, Jan 25, 2018 at 2:45 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I've been working on using FreeSWITCH as a bridge between different devices, and I need to delay all calls by 2-3 seconds in order for the external systems to keep up.
This delay should happen at the start of a call, and all subsequent audio should sound fine, just late. I know that doing this sounds silly - I'm sorry about that.
Here are my approaches:
1) Eavesdrop on a call send through delay_echo - this works for SIP clients but I have to do an absolute_codec_string as it seems that CODEC support is limited, and I'm often getting no delay at all with my PortAudio clients.
2) Use lua to do a session freeswitch.msleep on answer - exactly what Confluence tells me to avoid - if I sleep but don't consume audio, I think I can get the desired result.
3) Mess with the jitterbuffer - I haven't had any success with this yet.
4) Increase the buffer in mod_PortAudio.
Writing this out makes me sure that I'm missing something - a way of getting delay_echo to work for all calls, or a way to pause session media to get the desired effect.
Can you please help?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
-- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
Thank you for the reply. Here's what I tried: <action application="set" data="jitterbuffer_msec=3000:3200:20"/> Works for calls from SIP clients but not for PortAudio calls. Same with delay_echo - the CODEC is seen as RAW and so the jitterbuffer doesn't seem to be engaged (read_frame->packet doesn't have anything to work with so it skips the buffer). I have tried to have PortAudio > extension 1 > bridge to extension 2, which produces better results, but (in my hands at least) it causes memory leaks and is unwieldy. On Wed, Jan 24, 2018 at 11:59 PM, Alexey Sibyakin <alex@freeswitch.com> wrote:
JB should be fine. Any details of unsuccess? https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer
On Thu, Jan 25, 2018 at 2:45 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I've been working on using FreeSWITCH as a bridge between different devices, and I need to delay all calls by 2-3 seconds in order for the external systems to keep up.
This delay should happen at the start of a call, and all subsequent audio should sound fine, just late. I know that doing this sounds silly - I'm sorry about that.
Here are my approaches:
1) Eavesdrop on a call send through delay_echo - this works for SIP clients but I have to do an absolute_codec_string as it seems that CODEC support is limited, and I'm often getting no delay at all with my PortAudio clients.
2) Use lua to do a session freeswitch.msleep on answer - exactly what Confluence tells me to avoid - if I sleep but don't consume audio, I think I can get the desired result.
3) Mess with the jitterbuffer - I haven't had any success with this yet.
4) Increase the buffer in mod_PortAudio.
Writing this out makes me sure that I'm missing something - a way of getting delay_echo to work for all calls, or a way to pause session media to get the desired effect.
Can you please help?
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
--
Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Then you should try to set JB for b-leg via export nolocal. On Thu, Jan 25, 2018 at 3:44 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
Thank you for the reply. Here's what I tried:
<action application="set" data="jitterbuffer_msec=3000:3200:20"/>
Works for calls from SIP clients but not for PortAudio calls.
Same with delay_echo - the CODEC is seen as RAW and so the jitterbuffer doesn't seem to be engaged (read_frame->packet doesn't have anything to work with so it skips the buffer).
I have tried to have PortAudio > extension 1 > bridge to extension 2, which produces better results, but (in my hands at least) it causes memory leaks and is unwieldy.
On Wed, Jan 24, 2018 at 11:59 PM, Alexey Sibyakin <alex@freeswitch.com> wrote:
JB should be fine. Any details of unsuccess? https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer
On Thu, Jan 25, 2018 at 2:45 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I've been working on using FreeSWITCH as a bridge between different devices, and I need to delay all calls by 2-3 seconds in order for the external systems to keep up.
This delay should happen at the start of a call, and all subsequent audio should sound fine, just late. I know that doing this sounds silly - I'm sorry about that.
Here are my approaches:
1) Eavesdrop on a call send through delay_echo - this works for SIP clients but I have to do an absolute_codec_string as it seems that CODEC support is limited, and I'm often getting no delay at all with my PortAudio clients.
2) Use lua to do a session freeswitch.msleep on answer - exactly what Confluence tells me to avoid - if I sleep but don't consume audio, I think I can get the desired result.
3) Mess with the jitterbuffer - I haven't had any success with this yet.
4) Increase the buffer in mod_PortAudio.
Writing this out makes me sure that I'm missing something - a way of getting delay_echo to work for all calls, or a way to pause session media to get the desired effect.
Can you please help?
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
--
Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
-- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
also you'll have to set rtp_jitter_buffer_during_bridge to true. /b On Thu, Jan 25, 2018 at 2:22 AM, Alexey Sibyakin <alex@freeswitch.com> wrote:
Then you should try to set JB for b-leg via export nolocal.
On Thu, Jan 25, 2018 at 3:44 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
Thank you for the reply. Here's what I tried:
<action application="set" data="jitterbuffer_msec=3000:3200:20"/>
Works for calls from SIP clients but not for PortAudio calls.
Same with delay_echo - the CODEC is seen as RAW and so the jitterbuffer doesn't seem to be engaged (read_frame->packet doesn't have anything to work with so it skips the buffer).
I have tried to have PortAudio > extension 1 > bridge to extension 2, which produces better results, but (in my hands at least) it causes memory leaks and is unwieldy.
On Wed, Jan 24, 2018 at 11:59 PM, Alexey Sibyakin <alex@freeswitch.com> wrote:
JB should be fine. Any details of unsuccess? https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer
On Thu, Jan 25, 2018 at 2:45 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I've been working on using FreeSWITCH as a bridge between different devices, and I need to delay all calls by 2-3 seconds in order for the external systems to keep up.
This delay should happen at the start of a call, and all subsequent audio should sound fine, just late. I know that doing this sounds silly - I'm sorry about that.
Here are my approaches:
1) Eavesdrop on a call send through delay_echo - this works for SIP clients but I have to do an absolute_codec_string as it seems that CODEC support is limited, and I'm often getting no delay at all with my PortAudio clients.
2) Use lua to do a session freeswitch.msleep on answer - exactly what Confluence tells me to avoid - if I sleep but don't consume audio, I think I can get the desired result.
3) Mess with the jitterbuffer - I haven't had any success with this yet.
4) Increase the buffer in mod_PortAudio.
Writing this out makes me sure that I'm missing something - a way of getting delay_echo to work for all calls, or a way to pause session media to get the desired effect.
Can you please help?
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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/free switch-users http://www.freeswitch.org
--
Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
--
Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
-- Brian West | Co-founder and Developer Need Commercial support? email sales@freeswitch.com FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: brian@freeswitch.com Mobile: 918-424-9378 Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> [image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image: color-twitter-96.png] <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
I'm sorry, I'm just not having any luck getting this going. Ideally, I'd like to be able to originate the portaudio endpoint into park, and eavesdrop on it there, but I understand that's convoluted to work with, so as a test I have a single extension: <extension name="portaudio_USB"> <condition field="destination_number" expression="^1(\d{2})$"> <action application="answer"/> <action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_msec=3000:3200:20"/> <action application="bridge" data="portaudio/endpoints/USB_01"/> </condition> </extension> Dialing 101 from a SIP client (MicroSIP) as user 1000 from the default configuration causes me to be bridged to the portaudio endpoint (I can hear input from the channel), but there's no delay at all. On Thu, Jan 25, 2018 at 10:13 AM, Brian West <brian@freeswitch.com> wrote:
also you'll have to set rtp_jitter_buffer_during_bridge to true.
/b
On Thu, Jan 25, 2018 at 2:22 AM, Alexey Sibyakin <alex@freeswitch.com> wrote:
Then you should try to set JB for b-leg via export nolocal.
On Thu, Jan 25, 2018 at 3:44 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
Thank you for the reply. Here's what I tried:
<action application="set" data="jitterbuffer_msec=3000:3200:20"/>
Works for calls from SIP clients but not for PortAudio calls.
Same with delay_echo - the CODEC is seen as RAW and so the jitterbuffer doesn't seem to be engaged (read_frame->packet doesn't have anything to work with so it skips the buffer).
I have tried to have PortAudio > extension 1 > bridge to extension 2, which produces better results, but (in my hands at least) it causes memory leaks and is unwieldy.
On Wed, Jan 24, 2018 at 11:59 PM, Alexey Sibyakin <alex@freeswitch.com> wrote:
JB should be fine. Any details of unsuccess? https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer
On Thu, Jan 25, 2018 at 2:45 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I've been working on using FreeSWITCH as a bridge between different devices, and I need to delay all calls by 2-3 seconds in order for the external systems to keep up.
This delay should happen at the start of a call, and all subsequent audio should sound fine, just late. I know that doing this sounds silly - I'm sorry about that.
Here are my approaches:
1) Eavesdrop on a call send through delay_echo - this works for SIP clients but I have to do an absolute_codec_string as it seems that CODEC support is limited, and I'm often getting no delay at all with my PortAudio clients.
2) Use lua to do a session freeswitch.msleep on answer - exactly what Confluence tells me to avoid - if I sleep but don't consume audio, I think I can get the desired result.
3) Mess with the jitterbuffer - I haven't had any success with this yet.
4) Increase the buffer in mod_PortAudio.
Writing this out makes me sure that I'm missing something - a way of getting delay_echo to work for all calls, or a way to pause session media to get the desired effect.
Can you please help?
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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/free switch-users http://www.freeswitch.org
--
Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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/free switch-users http://www.freeswitch.org
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
--
Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378 <(918)%20424-9378>
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
[image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image: color-twitter-96.png] <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_msec=3000:3200:20"/> here you need to choose Alice or Bob will have JB, not both of them only half On Fri, Jan 26, 2018 at 12:35 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I'm sorry, I'm just not having any luck getting this going.
Ideally, I'd like to be able to originate the portaudio endpoint into park, and eavesdrop on it there, but I understand that's convoluted to work with, so as a test I have a single extension:
<extension name="portaudio_USB"> <condition field="destination_number" expression="^1(\d{2})$"> <action application="answer"/> <action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_ msec=3000:3200:20"/> <action application="bridge" data="portaudio/endpoints/USB_01"/> </condition> </extension>
Dialing 101 from a SIP client (MicroSIP) as user 1000 from the default configuration causes me to be bridged to the portaudio endpoint (I can hear input from the channel), but there's no delay at all.
On Thu, Jan 25, 2018 at 10:13 AM, Brian West <brian@freeswitch.com> wrote:
also you'll have to set rtp_jitter_buffer_during_bridge to true.
/b
On Thu, Jan 25, 2018 at 2:22 AM, Alexey Sibyakin <alex@freeswitch.com> wrote:
Then you should try to set JB for b-leg via export nolocal.
On Thu, Jan 25, 2018 at 3:44 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
Thank you for the reply. Here's what I tried:
<action application="set" data="jitterbuffer_msec=3000:3200:20"/>
Works for calls from SIP clients but not for PortAudio calls.
Same with delay_echo - the CODEC is seen as RAW and so the jitterbuffer doesn't seem to be engaged (read_frame->packet doesn't have anything to work with so it skips the buffer).
I have tried to have PortAudio > extension 1 > bridge to extension 2, which produces better results, but (in my hands at least) it causes memory leaks and is unwieldy.
On Wed, Jan 24, 2018 at 11:59 PM, Alexey Sibyakin <alex@freeswitch.com> wrote:
JB should be fine. Any details of unsuccess? https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer
On Thu, Jan 25, 2018 at 2:45 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I've been working on using FreeSWITCH as a bridge between different devices, and I need to delay all calls by 2-3 seconds in order for the external systems to keep up.
This delay should happen at the start of a call, and all subsequent audio should sound fine, just late. I know that doing this sounds silly - I'm sorry about that.
Here are my approaches:
1) Eavesdrop on a call send through delay_echo - this works for SIP clients but I have to do an absolute_codec_string as it seems that CODEC support is limited, and I'm often getting no delay at all with my PortAudio clients.
2) Use lua to do a session freeswitch.msleep on answer - exactly what Confluence tells me to avoid - if I sleep but don't consume audio, I think I can get the desired result.
3) Mess with the jitterbuffer - I haven't had any success with this yet.
4) Increase the buffer in mod_PortAudio.
Writing this out makes me sure that I'm missing something - a way of getting delay_echo to work for all calls, or a way to pause session media to get the desired effect.
Can you please help?
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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/free switch-users http://www.freeswitch.org
--
Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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/free switch-users http://www.freeswitch.org
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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/free switch-users http://www.freeswitch.org
--
Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378 <(918)%20424-9378>
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
[image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image: color-twitter-96.png] <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
-- Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
I've worked on this some more, and have been able to watch a little more of what's happening by splitting into two FreeSWITCH instances and sending traffic between them. I can make the jitter buffer and delay_echo provide the required delay, and this works fine with a basic SIP softphone. What's happening is that mod_portaudio is providing L16 at high sample rates and isn't transcoding it even across servers. It seems that I need to convert L16 48K to (say) PCMA or OPUS or something that delay_echo or the jitterbuffer could deal with. Is that something that's possible? I know I can eavesdrop or send to a conference, but is there a way I can easily convert from portaudio to something more generic to send to the dialplan? Erik On Sun, Jan 28, 2018 at 8:36 PM, Alexey Sibyakin <alex@freeswitch.com> wrote:
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_mse c=3000:3200:20"/>
here you need to choose Alice or Bob will have JB, not both of them only half
On Fri, Jan 26, 2018 at 12:35 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I'm sorry, I'm just not having any luck getting this going.
Ideally, I'd like to be able to originate the portaudio endpoint into park, and eavesdrop on it there, but I understand that's convoluted to work with, so as a test I have a single extension:
<extension name="portaudio_USB"> <condition field="destination_number" expression="^1(\d{2})$"> <action application="answer"/> <action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_mse c=3000:3200:20"/> <action application="bridge" data="portaudio/endpoints/USB_01"/> </condition> </extension>
Dialing 101 from a SIP client (MicroSIP) as user 1000 from the default configuration causes me to be bridged to the portaudio endpoint (I can hear input from the channel), but there's no delay at all.
On Thu, Jan 25, 2018 at 10:13 AM, Brian West <brian@freeswitch.com> wrote:
also you'll have to set rtp_jitter_buffer_during_bridge to true.
/b
On Thu, Jan 25, 2018 at 2:22 AM, Alexey Sibyakin <alex@freeswitch.com> wrote:
Then you should try to set JB for b-leg via export nolocal.
On Thu, Jan 25, 2018 at 3:44 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
Thank you for the reply. Here's what I tried:
<action application="set" data="jitterbuffer_msec=3000:3200:20"/>
Works for calls from SIP clients but not for PortAudio calls.
Same with delay_echo - the CODEC is seen as RAW and so the jitterbuffer doesn't seem to be engaged (read_frame->packet doesn't have anything to work with so it skips the buffer).
I have tried to have PortAudio > extension 1 > bridge to extension 2, which produces better results, but (in my hands at least) it causes memory leaks and is unwieldy.
On Wed, Jan 24, 2018 at 11:59 PM, Alexey Sibyakin <alex@freeswitch.com
wrote:
JB should be fine. Any details of unsuccess? https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer
On Thu, Jan 25, 2018 at 2:45 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
> I've been working on using FreeSWITCH as a bridge between different > devices, and I need to delay all calls by 2-3 seconds in order for the > external systems to keep up. > > This delay should happen at the start of a call, and all subsequent > audio should sound fine, just late. > I know that doing this sounds silly - I'm sorry about that. > > Here are my approaches: > > 1) Eavesdrop on a call send through delay_echo - this works for SIP > clients but I have to do an absolute_codec_string as it seems that CODEC > support is limited, and I'm often getting no delay at all with my PortAudio > clients. > > 2) Use lua to do a session freeswitch.msleep on answer - exactly > what Confluence tells me to avoid - if I sleep but don't consume audio, I > think I can get the desired result. > > 3) Mess with the jitterbuffer - I haven't had any success with this > yet. > > 4) Increase the buffer in mod_PortAudio. > > Writing this out makes me sure that I'm missing something - a way of > getting delay_echo to work for all calls, or a way to pause session media > to get the desired effect. > > Can you please help? > > ____________________________________________________________ > _____________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.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/free > switch-users > http://www.freeswitch.org >
--
Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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/free switch-users http://www.freeswitch.org
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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/free switch-users http://www.freeswitch.org
--
Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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/free switch-users http://www.freeswitch.org
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378 <(918)%20424-9378>
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
[image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image: color-twitter-96.png] <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
--
Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
OK, doing <action application="export" data="nolocal:absolute_codec_string=PCMA,PCMU"/> <action application="bridge" data="sofia/gateway/FreeSWITCH/201"/> Seems to take care of the transcoding. I'm going to try and move this back to one server. Erik On Mon, Jan 29, 2018 at 1:14 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I've worked on this some more, and have been able to watch a little more of what's happening by splitting into two FreeSWITCH instances and sending traffic between them.
I can make the jitter buffer and delay_echo provide the required delay, and this works fine with a basic SIP softphone.
What's happening is that mod_portaudio is providing L16 at high sample rates and isn't transcoding it even across servers.
It seems that I need to convert L16 48K to (say) PCMA or OPUS or something that delay_echo or the jitterbuffer could deal with.
Is that something that's possible? I know I can eavesdrop or send to a conference, but is there a way I can easily convert from portaudio to something more generic to send to the dialplan?
Erik
On Sun, Jan 28, 2018 at 8:36 PM, Alexey Sibyakin <alex@freeswitch.com> wrote:
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_mse c=3000:3200:20"/>
here you need to choose Alice or Bob will have JB, not both of them only half
On Fri, Jan 26, 2018 at 12:35 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I'm sorry, I'm just not having any luck getting this going.
Ideally, I'd like to be able to originate the portaudio endpoint into park, and eavesdrop on it there, but I understand that's convoluted to work with, so as a test I have a single extension:
<extension name="portaudio_USB"> <condition field="destination_number" expression="^1(\d{2})$"> <action application="answer"/> <action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_mse c=3000:3200:20"/> <action application="bridge" data="portaudio/endpoints/USB_01"/> </condition> </extension>
Dialing 101 from a SIP client (MicroSIP) as user 1000 from the default configuration causes me to be bridged to the portaudio endpoint (I can hear input from the channel), but there's no delay at all.
On Thu, Jan 25, 2018 at 10:13 AM, Brian West <brian@freeswitch.com> wrote:
also you'll have to set rtp_jitter_buffer_during_bridge to true.
/b
On Thu, Jan 25, 2018 at 2:22 AM, Alexey Sibyakin <alex@freeswitch.com> wrote:
Then you should try to set JB for b-leg via export nolocal.
On Thu, Jan 25, 2018 at 3:44 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
Thank you for the reply. Here's what I tried:
<action application="set" data="jitterbuffer_msec=3000:3200:20"/>
Works for calls from SIP clients but not for PortAudio calls.
Same with delay_echo - the CODEC is seen as RAW and so the jitterbuffer doesn't seem to be engaged (read_frame->packet doesn't have anything to work with so it skips the buffer).
I have tried to have PortAudio > extension 1 > bridge to extension 2, which produces better results, but (in my hands at least) it causes memory leaks and is unwieldy.
On Wed, Jan 24, 2018 at 11:59 PM, Alexey Sibyakin < alex@freeswitch.com> wrote:
> JB should be fine. Any details of unsuccess? > https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer > > On Thu, Jan 25, 2018 at 2:45 PM, Erik M. Devane - Comms Guy < > emdevane@gmail.com> wrote: > >> I've been working on using FreeSWITCH as a bridge between different >> devices, and I need to delay all calls by 2-3 seconds in order for the >> external systems to keep up. >> >> This delay should happen at the start of a call, and all subsequent >> audio should sound fine, just late. >> I know that doing this sounds silly - I'm sorry about that. >> >> Here are my approaches: >> >> 1) Eavesdrop on a call send through delay_echo - this works for SIP >> clients but I have to do an absolute_codec_string as it seems that CODEC >> support is limited, and I'm often getting no delay at all with my PortAudio >> clients. >> >> 2) Use lua to do a session freeswitch.msleep on answer - exactly >> what Confluence tells me to avoid - if I sleep but don't consume audio, I >> think I can get the desired result. >> >> 3) Mess with the jitterbuffer - I haven't had any success with this >> yet. >> >> 4) Increase the buffer in mod_PortAudio. >> >> Writing this out makes me sure that I'm missing something - a way >> of getting delay_echo to work for all calls, or a way to pause session >> media to get the desired effect. >> >> Can you please help? >> >> ____________________________________________________________ >> _____________ >> Professional FreeSWITCH Consulting Services: >> consulting@freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.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/free >> switch-users >> http://www.freeswitch.org >> > > > > -- > > Alex Sibyakin | Support Engineer > FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 > <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> > Email: alex@freeswitch.com > Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> > Need commercial support? Contact sales@freeswitch.com for details. > > ____________________________________________________________ > _____________ > Professional FreeSWITCH Consulting Services: > consulting@freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.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/free > switch-users > http://www.freeswitch.org >
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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/free switch-users http://www.freeswitch.org
--
Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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/free switch-users http://www.freeswitch.org
--
Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g>
Email: brian@freeswitch.com
Mobile: 918-424-9378 <(918)%20424-9378>
Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/>
[image: color-facebook-96.png] <https://www.facebook.com/freeswitch/>[image: color-twitter-96.png] <https://twitter.com/freeswitch?ref_src=twsrc%5Egoogle%7Ctwcamp%5Eserp%7Ctwgr%5Eauthor>
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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/free switch-users http://www.freeswitch.org
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
--
Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Hello, Other than the older AWS notes on Confluence, does anyone have any best practices for running FS successfully on AWS? I'm thinking of using the newer HVM virtualization. Can anyone share some specs for what they're running on AWS In terms of call volume, transcoding, etc? Thanks Mike
Bypass media if your usage model will allow it. On Mon, Jan 29, 2018 at 5:44 PM, Michael Avers <michael@mailworks.org> wrote:
Hello,
Other than the older AWS notes on Confluence, does anyone have any best practices for running FS successfully on AWS? I'm thinking of using the newer HVM virtualization.
Can anyone share some specs for what they're running on AWS In terms of call volume, transcoding, etc?
Thanks Mike
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
-- Sincerely, Russell Treleaven sip:rtreleaven@sip.bunnykick.ca
I am using server from AWS and its working without any problem. Just need to do NAT setting. Defined public IP in ext-rtp-ip and ext-sip-ip in profile. Nothing else required. I have used ASTPP Switch and IVR with voicemail transcription (using google API). Thanks Uday. On Tue, Jan 30, 2018 at 4:14 AM, Michael Avers <michael@mailworks.org> wrote:
Hello,
Other than the older AWS notes on Confluence, does anyone have any best practices for running FS successfully on AWS? I'm thinking of using the newer HVM virtualization.
Can anyone share some specs for what they're running on AWS In terms of call volume, transcoding, etc?
Thanks Mike
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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-- Thanks & Regard Uday. Mobile:- +91-9377579349
Since signalling is separate to media I suspect what you're doing here is swapping the 3 seconds of audio lost at the start of the call for 3 seconds of audio lost at the end of the call. I also assume this isn't an interactive call as a 3s latency is going to be almost unusable for a conversation. On 25 January 2018 at 06:44, Erik M. Devane - Comms Guy <emdevane@gmail.com> wrote:
Thank you for the reply. Here's what I tried:
<action application="set" data="jitterbuffer_msec=3000:3200:20"/>
Works for calls from SIP clients but not for PortAudio calls.
Same with delay_echo - the CODEC is seen as RAW and so the jitterbuffer doesn't seem to be engaged (read_frame->packet doesn't have anything to work with so it skips the buffer).
I have tried to have PortAudio > extension 1 > bridge to extension 2, which produces better results, but (in my hands at least) it causes memory leaks and is unwieldy.
On Wed, Jan 24, 2018 at 11:59 PM, Alexey Sibyakin <alex@freeswitch.com> wrote:
JB should be fine. Any details of unsuccess? https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer
On Thu, Jan 25, 2018 at 2:45 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I've been working on using FreeSWITCH as a bridge between different devices, and I need to delay all calls by 2-3 seconds in order for the external systems to keep up.
This delay should happen at the start of a call, and all subsequent audio should sound fine, just late. I know that doing this sounds silly - I'm sorry about that.
Here are my approaches:
1) Eavesdrop on a call send through delay_echo - this works for SIP clients but I have to do an absolute_codec_string as it seems that CODEC support is limited, and I'm often getting no delay at all with my PortAudio clients.
2) Use lua to do a session freeswitch.msleep on answer - exactly what Confluence tells me to avoid - if I sleep but don't consume audio, I think I can get the desired result.
3) Mess with the jitterbuffer - I haven't had any success with this yet.
4) Increase the buffer in mod_PortAudio.
Writing this out makes me sure that I'm missing something - a way of getting delay_echo to work for all calls, or a way to pause session media to get the desired effect.
Can you please help?
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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--
Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
That's right - I'm interfacing to a recorder system - my audio inputs are real-time but the metadata and routing information is sent AFTER the audio is received. By buffering a constant X seconds, I can make sure all of my information is ready before starting a conventional SIP/RTP session, and then just use sched_hangup to add the X back at the end. The complete chain (as seen by FreeSWITCH on a Windows Server) actually is: 64 channels of Dante > ASIO > PortAudio > Delay_Echo > Eavesdrop B-only > mod_managed originate from Eavesdrop to Park with mod_oreka inline in the dialplan to produce an RTP stream to a commercial, time stamped voice recorder. Is there a prize for most convoluted routing, or maybe most FS modules used? As I can't seem to get the jitterbuffer or delay_echo to work with 64K audio, I'm currently using one FS instance as a 64 channel softphone and another instance for talking to the third party devices. I do wish I could get the transcoding to happen on the same instance. On Wed, Jan 31, 2018 at 2:56 PM, Steven Ayre <steveayre@gmail.com> wrote:
Since signalling is separate to media I suspect what you're doing here is swapping the 3 seconds of audio lost at the start of the call for 3 seconds of audio lost at the end of the call. I also assume this isn't an interactive call as a 3s latency is going to be almost unusable for a conversation.
On 25 January 2018 at 06:44, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
Thank you for the reply. Here's what I tried:
<action application="set" data="jitterbuffer_msec=3000:3200:20"/>
Works for calls from SIP clients but not for PortAudio calls.
Same with delay_echo - the CODEC is seen as RAW and so the jitterbuffer doesn't seem to be engaged (read_frame->packet doesn't have anything to work with so it skips the buffer).
I have tried to have PortAudio > extension 1 > bridge to extension 2, which produces better results, but (in my hands at least) it causes memory leaks and is unwieldy.
On Wed, Jan 24, 2018 at 11:59 PM, Alexey Sibyakin <alex@freeswitch.com> wrote:
JB should be fine. Any details of unsuccess? https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer
On Thu, Jan 25, 2018 at 2:45 PM, Erik M. Devane - Comms Guy < emdevane@gmail.com> wrote:
I've been working on using FreeSWITCH as a bridge between different devices, and I need to delay all calls by 2-3 seconds in order for the external systems to keep up.
This delay should happen at the start of a call, and all subsequent audio should sound fine, just late. I know that doing this sounds silly - I'm sorry about that.
Here are my approaches:
1) Eavesdrop on a call send through delay_echo - this works for SIP clients but I have to do an absolute_codec_string as it seems that CODEC support is limited, and I'm often getting no delay at all with my PortAudio clients.
2) Use lua to do a session freeswitch.msleep on answer - exactly what Confluence tells me to avoid - if I sleep but don't consume audio, I think I can get the desired result.
3) Mess with the jitterbuffer - I haven't had any success with this yet.
4) Increase the buffer in mod_PortAudio.
Writing this out makes me sure that I'm missing something - a way of getting delay_echo to work for all calls, or a way to pause session media to get the desired effect.
Can you please help?
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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/free switch-users http://www.freeswitch.org
--
Alex Sibyakin | Support Engineer FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045 <https://maps.google.com/?q=17345+Civic+Drive+%232531+Brookfield,+WI+53045&entry=gmail&source=g> Email: alex@freeswitch.com Website: https://www.FreeSWITCH.com <https://www.freeswitch.com/> Need commercial support? Contact sales@freeswitch.com for details.
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.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 (7)
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Alexey Sibyakin
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Brian West
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Erik M. Devane - Comms Guy
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Michael Avers
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Russell Treleaven
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Steven Ayre
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Uday kumar