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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link="#0563C1" vlink="#954F72"><div class=WordSection1><p class=MsoNormal>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)<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>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?<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>I built FreeSWITCH Version 1.7.0+git~20160706T181946Z~8c6b2657bf~64bit (git 8c6b265 2016-07-06 18:19:46Z 64bit)<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>2016-08-03 19:34:43.901960 [DEBUG] switch_rtp.c:6711 Correct audio ip/port confirmed.<o:p></o:p></p><p class=MsoNormal>2016-08-03 19:34:43.901960 [WARNING] <b>switch_core_media.c:2568 [VBR]: Asynchronous PTIME supported, adjusting JB size. Remote PTIME changed from [20] to [36]</b><o:p></o:p></p><p class=MsoNormal>2016-08-03 19:34:43.921964 [NOTICE] switch_core_media.c:2977 Deactivating write resampler<o:p></o:p></p><p class=MsoNormal>2016-08-03 19:34:43.921964 [DEBUG] switch_core_media.c:2984 Changing Codec from SPEEX@20ms@16000hz to SPEEX@36ms@16000hz<o:p></o:p></p><p class=MsoNormal>2016-08-03 19:34:43.921964 [NOTICE] switch_core_io.c:1202 Activating write resampler<o:p></o:p></p><p class=MsoNormal>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<o:p></o:p></p><p class=MsoNormal>2016-08-03 19:34:43.961958 [ERR] switch_core_media.c:3021 Can't load codec?<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><b>The VBR is setting ptime to 36, 77, etc, varies every call coming in, which fails to find a match on speex implementation .<o:p></o:p></b></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Both opus and speex16 calls come in to echo, depending on if the browser is web-rtc capable to support fallback.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>We send send into echo, they press 1 to transfer here<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>root@ip-10-0-0-69:/opt/freeswitch/conf/dialplan/default# cat bbb_echo_test.xml<o:p></o:p></p><p class=MsoNormal><include><o:p></o:p></p><p class=MsoNormal> <extension name="bbb_echo_test_direct"><o:p></o:p></p><p class=MsoNormal> <condition field="${bbb_authorized}" expression="true" break="on-false"/><o:p></o:p></p><p class=MsoNormal> <condition field="destination_number" expression="^9196$|^9196(\d{5})$"><o:p></o:p></p><p class=MsoNormal> <action application="set" data="vbridge=$1"/><o:p></o:p></p><p class=MsoNormal> <action application="answer"/><o:p></o:p></p><p class=MsoNormal> <action application="bind_digit_action" data="direct_from_echo,1,exec:execute_extension,${vbridge} XML default"/><o:p></o:p></p><p class=MsoNormal> <action application="sleep" data="1500"/><o:p></o:p></p><p class=MsoNormal> <action application="echo"/><o:p></o:p></p><p class=MsoNormal> </condition><o:p></o:p></p><p class=MsoNormal> </extension><o:p></o:p></p><p class=MsoNormal></include><o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Then they are transferred.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>root@ip-10-0-0-69:/opt/freeswitch/conf/dialplan/default# cat bbb_conference.xml<o:p></o:p></p><p class=MsoNormal><include><o:p></o:p></p><p class=MsoNormal> <extension name="bbb_conferences"><o:p></o:p></p><p class=MsoNormal> <condition field="${bbb_authorized}" expression="true" break="on-false"/><o:p></o:p></p><p class=MsoNormal> <condition field="destination_number" expression="^(\d{5})$"><o:p></o:p></p><p class=MsoNormal> <b><action application="set" data="jitterbuffer_msec=20:400"/> <o:p></o:p></b></p><p class=MsoNormal> <action application="answer"/><o:p></o:p></p><p class=MsoNormal> <action application="conference" data="$1@cdquality"/><o:p></o:p></p><p class=MsoNormal> </condition><o:p></o:p></p><p class=MsoNormal> </extension><o:p></o:p></p><p class=MsoNormal></include><o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>So master from 7/06 currently after setting the jitterbuffer on speex call changes the PTIME to some number that doesn’t match.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Opus calls work fine.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>If I comment out the jitterbuffer in dialplan the calls work for both opus and speex.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Any help on how to get speex to stay fixed at 20ms like it had worked in previous with the jitterbuffer setting.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Can we set jitter buffer defaults for opus another way?<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Thanks for the help.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Regards,<o:p></o:p></p><p class=MsoNormal>Stephen<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>HostBBB – Online Learning Solutions <o:p></o:p></p><p class=MsoNormal>207 Technology Group Inc. 1-888-229-9756 skype: Stephen_Dame<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p></div></body></html>