[Freeswitch-users] [VBR]: Asynchronous PTIME supported on Speex16
Dragos Oancea
dragos.oancea at athonet.com
Fri Aug 19 19:39:36 MSD 2016
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 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 ||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 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.____
>
> __ __
>
>
> _________________________________________________________________________
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>
> --
>
> */Brian West/*
> brian at freeswitch.org <mailto:brian at freeswitch.org>
>
>
> */Twitter: @FreeSWITCH , @briankwest/*
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>
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