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

[Michael Jerris]

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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