[VBR]: Asynchronous PTIME supported on Speex16
Have an application that uses SPEEX@20ms@16000, everything works fine in 1.4, 1.5 and FreeSWITCH Version 1.7.0+git~20160219T153438Z~3bd26eaa6b~64bit (git 3bd26ea 2016-02-19 15:34:38Z 64bit) We bring a user into echo, then transfer them to a conference after they confirm they can hear themselves. We connect to audio fine at the 20ms and confirm. But the transfer is setting on VBR since updating freeswitch? I built FreeSWITCH Version 1.7.0+git~20160706T181946Z~8c6b2657bf~64bit (git 8c6b265 2016-07-06 18:19:46Z 64bit) 2016-08-03 19:34:43.901960 [DEBUG] switch_rtp.c:6711 Correct audio ip/port confirmed. 2016-08-03 19:34:43.901960 [WARNING] switch_core_media.c:2568 [VBR]: Asynchronous PTIME supported, adjusting JB size. Remote PTIME changed from [20] to [36] 2016-08-03 19:34:43.921964 [NOTICE] switch_core_media.c:2977 Deactivating write resampler 2016-08-03 19:34:43.921964 [DEBUG] switch_core_media.c:2984 Changing Codec from SPEEX@20ms@16000hz to SPEEX@36ms@16000hz 2016-08-03 19:34:43.921964 [NOTICE] switch_core_io.c:1202 Activating write resampler 2016-08-03 19:34:43.961958 [WARNING] switch_core_codec.c:721 Codec SPEEX Exists but not at the desired implementation. 16000hz 36ms 1ch 2016-08-03 19:34:43.961958 [ERR] switch_core_media.c:3021 Can't load codec? The VBR is setting ptime to 36, 77, etc, varies every call coming in, which fails to find a match on speex implementation . Both opus and speex16 calls come in to echo, depending on if the browser is web-rtc capable to support fallback. We send send into echo, they press 1 to transfer here root@ip-10-0-0-69:/opt/freeswitch/conf/dialplan/default# cat bbb_echo_test.xml <include> <extension name="bbb_echo_test_direct"> <condition field="${bbb_authorized}" expression="true" break="on-false"/> <condition field="destination_number" expression="^9196$|^9196(\d{5})$"> <action application="set" data="vbridge=$1"/> <action application="answer"/> <action application="bind_digit_action" data="direct_from_echo,1,exec:execute_extension,${vbridge} XML default"/> <action application="sleep" data="1500"/> <action application="echo"/> </condition> </extension> </include> Then they are transferred. root@ip-10-0-0-69:/opt/freeswitch/conf/dialplan/default# cat bbb_conference.xml <include> <extension name="bbb_conferences"> <condition field="${bbb_authorized}" expression="true" break="on-false"/> <condition field="destination_number" expression="^(\d{5})$"> <action application="set" data="jitterbuffer_msec=20:400"/> <action application="answer"/> <action application="conference" data="$1@cdquality"/> </condition> </extension> </include> So master from 7/06 currently after setting the jitterbuffer on speex call changes the PTIME to some number that doesn't match. Opus calls work fine. If I comment out the jitterbuffer in dialplan the calls work for both opus and speex. Any help on how to get speex to stay fixed at 20ms like it had worked in previous with the jitterbuffer setting. Can we set jitter buffer defaults for opus another way? Thanks for the help. Regards, Stephen HostBBB - Online Learning Solutions 207 Technology Group Inc. 1-888-229-9756 skype: Stephen_Dame
Try current master, we did some work on this and i think its correct now.
On Aug 3, 2016, at 4:40 PM, Stephen Dame <sdame@207me.com> wrote:
Have an application that uses SPEEX@20ms@16000, everything works fine in 1.4, 1.5 and FreeSWITCH Version 1.7.0+git~20160219T153438Z~3bd26eaa6b~64bit (git 3bd26ea 2016-02-19 15:34:38Z 64bit)
We bring a user into echo, then transfer them to a conference after they confirm they can hear themselves. We connect to audio fine at the 20ms and confirm. But the transfer is setting on VBR since updating freeswitch?
I built FreeSWITCH Version 1.7.0+git~20160706T181946Z~8c6b2657bf~64bit (git 8c6b265 2016-07-06 18:19:46Z 64bit)
2016-08-03 19:34:43.901960 [DEBUG] switch_rtp.c:6711 Correct audio ip/port confirmed. 2016-08-03 19:34:43.901960 [WARNING] switch_core_media.c:2568 [VBR]: Asynchronous PTIME supported, adjusting JB size. Remote PTIME changed from [20] to [36] 2016-08-03 19:34:43.921964 [NOTICE] switch_core_media.c:2977 Deactivating write resampler 2016-08-03 19:34:43.921964 [DEBUG] switch_core_media.c:2984 Changing Codec from SPEEX@20ms@16000hz to SPEEX@36ms@16000hz 2016-08-03 19:34:43.921964 [NOTICE] switch_core_io.c:1202 Activating write resampler 2016-08-03 19:34:43.961958 [WARNING] switch_core_codec.c:721 Codec SPEEX Exists but not at the desired implementation. 16000hz 36ms 1ch 2016-08-03 19:34:43.961958 [ERR] switch_core_media.c:3021 Can't load codec?
The VBR is setting ptime to 36, 77, etc, varies every call coming in, which fails to find a match on speex implementation .
Both opus and speex16 calls come in to echo, depending on if the browser is web-rtc capable to support fallback.
We send send into echo, they press 1 to transfer here
root@ip-10-0-0-69:/opt/freeswitch/conf/dialplan/default# cat bbb_echo_test.xml <include> <extension name="bbb_echo_test_direct"> <condition field="${bbb_authorized}" expression="true" break="on-false"/> <condition field="destination_number" expression="^9196$|^9196(\d{5})$"> <action application="set" data="vbridge=$1"/> <action application="answer"/> <action application="bind_digit_action" data="direct_from_echo,1,exec:execute_extension,${vbridge} XML default"/> <action application="sleep" data="1500"/> <action application="echo"/> </condition> </extension> </include>
Then they are transferred.
root@ip-10-0-0-69:/opt/freeswitch/conf/dialplan/default# cat bbb_conference.xml <include> <extension name="bbb_conferences"> <condition field="${bbb_authorized}" expression="true" break="on-false"/> <condition field="destination_number" expression="^(\d{5})$"> <action application="set" data="jitterbuffer_msec=20:400"/> <action application="answer"/> <action application="conference" data="$1@cdquality"/> </condition> </extension> </include>
So master from 7/06 currently after setting the jitterbuffer on speex call changes the PTIME to some number that doesn’t match.
Opus calls work fine.
If I comment out the jitterbuffer in dialplan the calls work for both opus and speex.
Any help on how to get speex to stay fixed at 20ms like it had worked in previous with the jitterbuffer setting.
Can we set jitter buffer defaults for opus another way?
Thanks for the help.
Mike, thanks I’ll give master a try this evening. Is there a way to set <action application="set" data="jitterbuffer_msec=20:400"/> as default for just any opus call, and not in this dialplan. I see I can set <param name="auto-jitterbuffer-msec" value="60"/> in profile, I can try that but assume it will heave the same. Trying to temporarily fix this since we live on the bleeding edge and have it in production :) Or maybe look at a channel variable to determine is it’s an opus call or speex and only set jitter for opus calls for now. The fallback to flash is rarely used, so missed this one in testing. Thanks again. Regards, Stephen HostBBB – Online Learning Solutions 207 Technology Group Inc. 1-888-229-9756 skype: Stephen_Dame From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Wednesday, August 03, 2016 6:07 PM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] [VBR]: Asynchronous PTIME supported on Speex16 Try current master, we did some work on this and i think its correct now. On Aug 3, 2016, at 4:40 PM, Stephen Dame <sdame@207me.com <mailto:sdame@207me.com> > wrote: Have an application that uses SPEEX@20ms@16000, everything works fine in 1.4, 1.5 and FreeSWITCH Version 1.7.0+git~20160219T153438Z~3bd26eaa6b~64bit (git 3bd26ea 2016-02-19 15:34:38Z 64bit) We bring a user into echo, then transfer them to a conference after they confirm they can hear themselves. We connect to audio fine at the 20ms and confirm. But the transfer is setting on VBR since updating freeswitch? I built FreeSWITCH Version 1.7.0+git~20160706T181946Z~8c6b2657bf~64bit (git 8c6b265 2016-07-06 18:19:46Z 64bit) 2016-08-03 19:34:43.901960 [DEBUG] switch_rtp.c:6711 Correct audio ip/port confirmed. 2016-08-03 19:34:43.901960 [WARNING] switch_core_media.c:2568 [VBR]: Asynchronous PTIME supported, adjusting JB size. Remote PTIME changed from [20] to [36] 2016-08-03 19:34:43.921964 [NOTICE] switch_core_media.c:2977 Deactivating write resampler 2016-08-03 19:34:43.921964 [DEBUG] switch_core_media.c:2984 Changing Codec from SPEEX@20ms@16000hz to SPEEX@36ms@16000hz 2016-08-03 19:34:43.921964 [NOTICE] switch_core_io.c:1202 Activating write resampler 2016-08-03 19:34:43.961958 [WARNING] switch_core_codec.c:721 Codec SPEEX Exists but not at the desired implementation. 16000hz 36ms 1ch 2016-08-03 19:34:43.961958 [ERR] switch_core_media.c:3021 Can't load codec? The VBR is setting ptime to 36, 77, etc, varies every call coming in, which fails to find a match on speex implementation . Both opus and speex16 calls come in to echo, depending on if the browser is web-rtc capable to support fallback. We send send into echo, they press 1 to transfer here root@ip-10-0-0-69:/opt/freeswitch/conf/dialplan/default# cat bbb_echo_test.xml <include> <extension name="bbb_echo_test_direct"> <condition field="${bbb_authorized}" expression="true" break="on-false"/> <condition field="destination_number" expression="^9196$|^9196(\d{5})$"> <action application="set" data="vbridge=$1"/> <action application="answer"/> <action application="bind_digit_action" data="direct_from_echo,1,exec:execute_extension,${vbridge} XML default"/> <action application="sleep" data="1500"/> <action application="echo"/> </condition> </extension> </include> Then they are transferred. root@ip-10-0-0-69:/opt/freeswitch/conf/dialplan/default# cat bbb_conference.xml <include> <extension name="bbb_conferences"> <condition field="${bbb_authorized}" expression="true" break="on-false"/> <condition field="destination_number" expression="^(\d{5})$"> <action application="set" data="jitterbuffer_msec=20:400"/> <action application="answer"/> <action application="conference" data="$1@cdquality"/> </condition> </extension> </include> So master from 7/06 currently after setting the jitterbuffer on speex call changes the PTIME to some number that doesn’t match. Opus calls work fine. If I comment out the jitterbuffer in dialplan the calls work for both opus and speex. Any help on how to get speex to stay fixed at 20ms like it had worked in previous with the jitterbuffer setting. Can we set jitter buffer defaults for opus another way? Thanks for the help.
You also never filed a JIRA did you? Remember if its not in JIRA we'll lose track of it. On Wed, Aug 3, 2016 at 5:42 PM, Stephen Dame <sdame@207me.com> wrote:
Mike, thanks I’ll give master a try this evening.
Is there a way to set <action application="set" data="jitterbuffer_msec=20:400"/> as default for just any opus call, and not in this dialplan.
I see I can set <param name="auto-jitterbuffer-msec" value="60"/> in profile, I can try that but assume it will heave the same.
Trying to temporarily fix this since we live on the bleeding edge and have it in production J
Or maybe look at a channel variable to determine is it’s an opus call or speex and only set jitter for opus calls for now.
The fallback to flash is rarely used, so missed this one in testing.
Thanks again.
Regards,
Stephen
HostBBB – Online Learning Solutions
207 Technology Group Inc. 1-888-229-9756 skype: Stephen_Dame
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Michael Jerris *Sent:* Wednesday, August 03, 2016 6:07 PM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> *Subject:* Re: [Freeswitch-users] [VBR]: Asynchronous PTIME supported on Speex16
Try current master, we did some work on this and i think its correct now.
On Aug 3, 2016, at 4:40 PM, Stephen Dame <sdame@207me.com> wrote:
Have an application that uses SPEEX@20ms@16000, everything works fine in 1.4, 1.5 and FreeSWITCH Version 1.7.0+git~20160219T153438Z~3bd26eaa6b~64bit (git 3bd26ea 2016-02-19 15:34:38Z 64bit)
We bring a user into echo, then transfer them to a conference after they confirm they can hear themselves. We connect to audio fine at the 20ms and confirm. But the transfer is setting on VBR since updating freeswitch?
I built FreeSWITCH Version 1.7.0+git~20160706T181946Z~8c6b2657bf~64bit (git 8c6b265 2016-07-06 18:19:46Z 64bit)
2016-08-03 19:34:43.901960 [DEBUG] switch_rtp.c:6711 Correct audio ip/port confirmed.
2016-08-03 19:34:43.901960 [WARNING] *switch_core_media.c:2568 [VBR]: Asynchronous PTIME supported, adjusting JB size. Remote PTIME changed from [20] to [36]*
2016-08-03 19:34:43.921964 [NOTICE] switch_core_media.c:2977 Deactivating write resampler
2016-08-03 19:34:43.921964 [DEBUG] switch_core_media.c:2984 Changing Codec from SPEEX@20ms@16000hz to SPEEX@36ms@16000hz
2016-08-03 19:34:43.921964 [NOTICE] switch_core_io.c:1202 Activating write resampler
2016-08-03 19:34:43.961958 [WARNING] switch_core_codec.c:721 Codec SPEEX Exists but not at the desired implementation. 16000hz 36ms 1ch
2016-08-03 19:34:43.961958 [ERR] switch_core_media.c:3021 Can't load codec?
*The VBR is setting ptime to 36, 77, etc, varies every call coming in, which fails to find a match on speex implementation .*
Both opus and speex16 calls come in to echo, depending on if the browser is web-rtc capable to support fallback.
We send send into echo, they press 1 to transfer here
root@ip-10-0-0-69:/opt/freeswitch/conf/dialplan/default# cat bbb_echo_test.xml
<include>
<extension name="bbb_echo_test_direct">
<condition field="${bbb_authorized}" expression="true" break="on-false"/>
<condition field="destination_number" expression="^9196$|^9196(\d{5} )$">
<action application="set" data="vbridge=$1"/>
<action application="answer"/>
<action application="bind_digit_action" data="direct_from_echo,1,exec:execute_extension,${vbridge} XML default"/>
<action application="sleep" data="1500"/>
<action application="echo"/>
</condition>
</extension>
</include>
Then they are transferred.
root@ip-10-0-0-69:/opt/freeswitch/conf/dialplan/default# cat bbb_conference.xml
<include>
<extension name="bbb_conferences">
<condition field="${bbb_authorized}" expression="true" break="on-false"/>
<condition field="destination_number" expression="^(\d{5})$">
*<action application="set" data="jitterbuffer_msec=20:400"/> *
<action application="answer"/>
<action application="conference" data="$1@cdquality"/>
</condition>
</extension>
</include>
So master from 7/06 currently after setting the jitterbuffer on speex call changes the PTIME to some number that doesn’t match.
Opus calls work fine.
If I comment out the jitterbuffer in dialplan the calls work for both opus and speex.
Any help on how to get speex to stay fixed at 20ms like it had worked in previous with the jitterbuffer setting.
Can we set jitter buffer defaults for opus another way?
Thanks for the 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
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-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
Hello, Try <param name="rtp-autofix-timing" value="false"/> That bug entered in connection with FS-8583 , it's something that was not tested enough, I only had a custom client that changed ptime by itself but I did not simulate very much with jitter and packet loss and with that exact same change. The original proposal was not to reset the codec, and it was only tested with Opus. The whole idea was to increase/decrease the JB size if the remote ptime changes, not to deal with reinitializing codecs. For example Opus can decode a 40 ms audio packet even if it's initialized with 20 ms. Not sure Speex could do that. Cheers, Dragos On 19/08/2016 16:47, Brian West wrote:
You also never filed a JIRA did you? Remember if its not in JIRA we'll lose track of it.
On Wed, Aug 3, 2016 at 5:42 PM, Stephen Dame <sdame@207me.com <mailto:sdame@207me.com>> wrote:
Mike, thanks I’ll give master a try this evening.____
__ __
Is there a way to set <action application="set" data="jitterbuffer_msec=20:400"/> as default for just any opus call, and not in this dialplan.____
__ __
I see I can set |<param| |name="auto-jitterbuffer-msec"| |value="60"/> in profile, I can try that but assume it will heave the same.____|
|__ __|
|Trying to temporarily fix this since we live on the bleeding edge and have it in production ||J||____|
|__ __|
|Or maybe look at a channel variable to determine is it’s an opus call or speex and only set jitter for opus calls for now.____|
|__ __|
|The fallback to flash is rarely used, so missed this one in testing.____|
|__ __|
|Thanks again.____|
__ __
Regards,____
Stephen____
__ __
HostBBB – Online Learning Solutions ____
207 Technology Group Inc. 1-888-229-9756 <tel:1-888-229-9756> skype: Stephen_Dame____
__ __
*From:*freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org>] *On Behalf Of *Michael Jerris *Sent:* Wednesday, August 03, 2016 6:07 PM *To:* FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> *Subject:* Re: [Freeswitch-users] [VBR]: Asynchronous PTIME supported on Speex16____
__ __
Try current master, we did some work on this and i think its correct now.____
__ __
On Aug 3, 2016, at 4:40 PM, Stephen Dame <sdame@207me.com <mailto:sdame@207me.com>> wrote:____
__ __
Have an application that uses SPEEX@20ms@16000, everything works fine in 1.4, 1.5 and FreeSWITCH Version 1.7.0+git~20160219T153438Z~3bd26eaa6b~64bit (git 3bd26ea 2016-02-19 15:34:38Z 64bit)____
____
We bring a user into echo, then transfer them to a conference after they confirm they can hear themselves. We connect to audio fine at the 20ms and confirm. But the transfer is setting on VBR since updating freeswitch?____
____
I built FreeSWITCH Version 1.7.0+git~20160706T181946Z~8c6b2657bf~64bit (git 8c6b265 2016-07-06 18:19:46Z 64bit)____
____
2016-08-03 19:34:43.901960 [DEBUG] switch_rtp.c:6711 Correct audio ip/port confirmed.____
2016-08-03 19:34:43.901960 [WARNING] *switch_core_media.c:2568 [VBR]: Asynchronous PTIME supported, adjusting JB size. Remote PTIME changed from [20] to [36]*____
2016-08-03 19:34:43.921964 [NOTICE] switch_core_media.c:2977 Deactivating write resampler____
2016-08-03 19:34:43.921964 [DEBUG] switch_core_media.c:2984 Changing Codec from SPEEX@20ms@16000hz to SPEEX@36ms@16000hz____
2016-08-03 19:34:43.921964 [NOTICE] switch_core_io.c:1202 Activating write resampler____
2016-08-03 19:34:43.961958 [WARNING] switch_core_codec.c:721 Codec SPEEX Exists but not at the desired implementation. 16000hz 36ms 1ch____
2016-08-03 19:34:43.961958 [ERR] switch_core_media.c:3021 Can't load codec?____
____
*The VBR is setting ptime to 36, 77, etc, varies every call coming in, which fails to find a match on speex implementation .*____
____
Both opus and speex16 calls come in to echo, depending on if the browser is web-rtc capable to support fallback.____
____
We send send into echo, they press 1 to transfer here____
____
root@ip-10-0-0-69:/opt/freeswitch/conf/dialplan/default# cat bbb_echo_test.xml____
<include>____
<extension name="bbb_echo_test_direct">____
<condition field="${bbb_authorized}" expression="true" break="on-false"/>____
<condition field="destination_number" expression="^9196$|^9196(\d{5})$">____
<action application="set" data="vbridge=$1"/>____
<action application="answer"/>____
<action application="bind_digit_action" data="direct_from_echo,1,exec:execute_extension,${vbridge} XML default"/>____
<action application="sleep" data="1500"/>____
<action application="echo"/>____
</condition>____
</extension>____
</include>____
____
Then they are transferred.____
____
root@ip-10-0-0-69:/opt/freeswitch/conf/dialplan/default# cat bbb_conference.xml____
<include>____
<extension name="bbb_conferences">____
<condition field="${bbb_authorized}" expression="true" break="on-false"/>____
<condition field="destination_number" expression="^(\d{5})$">____
*<action application="set" data="jitterbuffer_msec=20:400"/> *____
<action application="answer"/>____
<action application="conference" data="$1@cdquality"/>____
</condition>____
</extension>____
</include>____
____
____
So master from 7/06 currently after setting the jitterbuffer on speex call changes the PTIME to some number that doesn’t match.____
____
Opus calls work fine.____
____
If I comment out the jitterbuffer in dialplan the calls work for both opus and speex.____
____
Any help on how to get speex to stay fixed at 20ms like it had worked in previous with the jitterbuffer setting.____
____
Can we set jitter buffer defaults for opus another way?____
____
Thanks for the help.____
__ __
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com>
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org <http://confluence.freeswitch.org> http://www.cluecon.com
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--
*/Brian West/* brian@freeswitch.org <mailto:brian@freeswitch.org>
*/Twitter: @FreeSWITCH , @briankwest/* http://www.freeswitchbook.com http://www.freeswitchcookbook.com https://www.gofundme.com/freeswitch_ubuntu
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest
_________________________________________________________________________ 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
Hi Brian, I was able to just modify dialplan, and not set jitter with speex16 fallback, only with the webrtc calls. Mike thought is was already fixed in master, I have not had a chance to circle around and confirm this. Let me try this week. If it’s still and issue will file jira. Regards, Stephen HostBBB – Online Learning Solutions 207 Technology Group Inc. 1-888-229-9756 skype: Stephen_Dame From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, August 19, 2016 10:48 AM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] [VBR]: Asynchronous PTIME supported on Speex16 You also never filed a JIRA did you? Remember if its not in JIRA we'll lose track of it. On Wed, Aug 3, 2016 at 5:42 PM, Stephen Dame <sdame@207me.com <mailto:sdame@207me.com> > wrote: Mike, thanks I’ll give master a try this evening. Is there a way to set <action application="set" data="jitterbuffer_msec=20:400"/> as default for just any opus call, and not in this dialplan. I see I can set <param name="auto-jitterbuffer-msec" value="60"/> in profile, I can try that but assume it will heave the same. Trying to temporarily fix this since we live on the bleeding edge and have it in production :) Or maybe look at a channel variable to determine is it’s an opus call or speex and only set jitter for opus calls for now. The fallback to flash is rarely used, so missed this one in testing. Thanks again. Regards, Stephen HostBBB – Online Learning Solutions 207 Technology Group Inc. 1-888-229-9756 <tel:1-888-229-9756> skype: Stephen_Dame From: freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org <mailto:freeswitch-users-bounces@lists.freeswitch.org> ] On Behalf Of Michael Jerris Sent: Wednesday, August 03, 2016 6:07 PM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> > Subject: Re: [Freeswitch-users] [VBR]: Asynchronous PTIME supported on Speex16 Try current master, we did some work on this and i think its correct now. On Aug 3, 2016, at 4:40 PM, Stephen Dame <sdame@207me.com <mailto:sdame@207me.com> > wrote: Have an application that uses SPEEX@20ms@16000, everything works fine in 1.4, 1.5 and FreeSWITCH Version 1.7.0+git~20160219T153438Z~3bd26eaa6b~64bit (git 3bd26ea 2016-02-19 15:34:38Z 64bit) We bring a user into echo, then transfer them to a conference after they confirm they can hear themselves. We connect to audio fine at the 20ms and confirm. But the transfer is setting on VBR since updating freeswitch? I built FreeSWITCH Version 1.7.0+git~20160706T181946Z~8c6b2657bf~64bit (git 8c6b265 2016-07-06 18:19:46Z 64bit) 2016-08-03 19:34:43.901960 [DEBUG] switch_rtp.c:6711 Correct audio ip/port confirmed. 2016-08-03 19:34:43.901960 [WARNING] switch_core_media.c:2568 [VBR]: Asynchronous PTIME supported, adjusting JB size. Remote PTIME changed from [20] to [36] 2016-08-03 19:34:43.921964 [NOTICE] switch_core_media.c:2977 Deactivating write resampler 2016-08-03 19:34:43.921964 [DEBUG] switch_core_media.c:2984 Changing Codec from SPEEX@20ms@16000hz to SPEEX@36ms@16000hz 2016-08-03 19:34:43.921964 [NOTICE] switch_core_io.c:1202 Activating write resampler 2016-08-03 19:34:43.961958 [WARNING] switch_core_codec.c:721 Codec SPEEX Exists but not at the desired implementation. 16000hz 36ms 1ch 2016-08-03 19:34:43.961958 [ERR] switch_core_media.c:3021 Can't load codec? The VBR is setting ptime to 36, 77, etc, varies every call coming in, which fails to find a match on speex implementation . Both opus and speex16 calls come in to echo, depending on if the browser is web-rtc capable to support fallback. We send send into echo, they press 1 to transfer here root@ip-10-0-0-69:/opt/freeswitch/conf/dialplan/default# cat bbb_echo_test.xml <include> <extension name="bbb_echo_test_direct"> <condition field="${bbb_authorized}" expression="true" break="on-false"/> <condition field="destination_number" expression="^9196$|^9196(\d{5})$"> <action application="set" data="vbridge=$1"/> <action application="answer"/> <action application="bind_digit_action" data="direct_from_echo,1,exec:execute_extension,${vbridge} XML default"/> <action application="sleep" data="1500"/> <action application="echo"/> </condition> </extension> </include> Then they are transferred. root@ip-10-0-0-69:/opt/freeswitch/conf/dialplan/default# cat bbb_conference.xml <include> <extension name="bbb_conferences"> <condition field="${bbb_authorized}" expression="true" break="on-false"/> <condition field="destination_number" expression="^(\d{5})$"> <action application="set" data="jitterbuffer_msec=20:400"/> <action application="answer"/> <action application="conference" data="$1@cdquality"/> </condition> </extension> </include> So master from 7/06 currently after setting the jitterbuffer on speex call changes the PTIME to some number that doesn’t match. Opus calls work fine. If I comment out the jitterbuffer in dialplan the calls work for both opus and speex. Any help on how to get speex to stay fixed at 20ms like it had worked in previous with the jitterbuffer setting. Can we set jitter buffer defaults for opus another way? Thanks for the help. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto: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 <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West brian@freeswitch.org <mailto:brian@freeswitch.org> <http://billing.freeswitch.org/templates/default/img/whmcslogo.png> Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com <https://www.gofundme.com/freeswitch_ubuntu> https://www.gofundme.com/freeswitch_ubuntu Got Bugs? 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participants (4)
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Brian West
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Dragos Oancea
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Michael Jerris
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Stephen Dame