[Freeswitch-users] [VBR]: Asynchronous PTIME supported on Speex16

Fri Aug 19 18:47:34 MSD 2016

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 at 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 at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Jerris
> *Sent:* Wednesday, August 03, 2016 6:07 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at 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 at 207me.com> wrote:
>
>
>
> Have an application that uses SPEEX at 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 at 20ms@16000hz to SPEEX at 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 at 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 at 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 at 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.
>
>
>
> _________________________________________________________________________
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-- 

*Brian West*
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