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. Regards, Stephen HostBBB - Online Learning Solutions 207 Technology Group Inc. 1-888-229-9756 skype: Stephen_Dame