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

Mon Aug 22 16:35:37 MSD 2016

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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Friday, August 19, 2016 10:48 AM
To: FreeSWITCH Users Help <freeswitch-users at 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 at 207me.com <mailto: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 :)

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 at lists.freeswitch.org <mailto:freeswitch-users-bounces at lists.freeswitch.org>  [mailto: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 <mailto: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 <mailto: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.

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at 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 at lists.freeswitch.org <mailto:FreeSWITCH-users at 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 at freeswitch.org <mailto:brian at 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? 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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160822/2ad6a640/attachment-0001.html 