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.

 

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