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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif'>If you are limiting the calls to specific codecs and avoiding transcoding, proxy media doesn’t really reduce the overhead anymore… that changed a few years ago but the notion its better still hangs on today<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri",sans-serif'><o:p> </o:p></span></p><p class=MsoNormal><b><span style='font-size:11.0pt;font-family:"Calibri",sans-serif'>From:</span></b><span style='font-size:11.0pt;font-family:"Calibri",sans-serif'> freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] <b>On Behalf Of </b>L?i Ð?ng<br><b>Sent:</b> Tuesday, November 22, 2016 9:07 PM<br><b>To:</b> FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org><br><b>Subject:</b> Re: [Freeswitch-users] absolute_codec_string not working<o:p></o:p></span></p><p class=MsoNormal><o:p> </o:p></p><div><div><p class=MsoNormal style='margin-bottom:12.0pt'>Hi @Michael, you were right, I'm intentionally using media_proxy for FS, since I want to reduce CPU usage on FS machine.<o:p></o:p></p></div><div><p class=MsoNormal>In this case, I just want to limit the codecs used for each endpoint, and codec negotiation will be handled by them.<o:p></o:p></p></div><p class=MsoNormal>e.g: caller use PCMA, PCMU, GSM by its own in INVITE, I want to limit the callee to only use PCMA,GSM.<br><br>Look like `absolute_codec_string` is not what I'm looking for right? Any way out?<o:p></o:p></p><div><p class=MsoNormal><br clear=all><o:p></o:p></p><div><div><div><div><div><div><div><p class=MsoNormal>Loi Dang Thanh<o:p></o:p></p></div><p class=MsoNormal>Phone : 01224.735.448<o:p></o:p></p></div><p class=MsoNormal>Email : <a href="mailto:loi.dangthanh@gmail.com" target="_blank">loi.dangthanh@gmail.com</a><o:p></o:p></p></div></div></div></div></div><p class=MsoNormal><o:p> </o:p></p><div><p class=MsoNormal>On Tue, Nov 22, 2016 at 9:57 PM, Michael Jerris <<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>> wrote:<o:p></o:p></p><blockquote style='border:none;border-left:solid #CCCCCC 1.0pt;padding:0in 0in 0in 6.0pt;margin-left:4.8pt;margin-right:0in'><div><p class=MsoNormal>using proxy_media is my best guess but can’t tell with this little info.<o:p></o:p></p><div><p class=MsoNormal><o:p> </o:p></p><div><blockquote style='margin-top:5.0pt;margin-bottom:5.0pt'><div><div><div><p class=MsoNormal>On Nov 22, 2016, at 5:27 AM, Lợi Đặng <<a href="mailto:loi.dangthanh@gmail.com" target="_blank">loi.dangthanh@gmail.com</a>> wrote:<o:p></o:p></p></div><p class=MsoNormal><o:p> </o:p></p></div></div><div><div><div><div><div><div><div><p class=MsoNormal><o:p> </o:p></p></div><p class=MsoNormal>Hi List, I got some trouble with using `absolute_codec_string` param.<o:p></o:p></p></div><p class=MsoNormal>My call scenario is pretty simple: caller <--> FS <--> callee.<o:p></o:p></p></div><p class=MsoNormal>My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I'm doing `<action application="bridge" data="{absolute_codec_string=PCMU,GSM}sofia/gateway/callee/$1"/>` in the dialplan.<o:p></o:p></p><div><p class=MsoNormal>But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the callee.<o:p></o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'>not sure what I'm missing, helps would be appreciated.<o:p></o:p></p></div><div><p class=MsoNormal>Note that when I'm using `originate` application in fs_cli, things are good.<o:p></o:p></p></div><div><div><div><div><p class=MsoNormal>`originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. <o:p></o:p></p></div><div><p class=MsoNormal style='margin-bottom:12.0pt'>I have FS with proper behavior in transcoding, caller has `m=audio 31184 RTP/AVP 0 101` received, and callee has `m=audio 21922 RTP/AVP 8 101` received.<o:p></o:p></p></div><div><p class=MsoNormal>rgds,<br clear=all><o:p></o:p></p></div><div><div><div><div><div><div><div><div><p class=MsoNormal>Loi Dang Thanh<o:p></o:p></p></div><p class=MsoNormal>Phone : 84.1224.735.448<o:p></o:p></p></div><p class=MsoNormal>Email : <a href="mailto:loi.dangthanh@gmail.com" target="_blank">loi.dangthanh@gmail.com</a><o:p></o:p></p></div></div></div></div></div></div></div></div></div></div></div></div><p class=MsoNormal>_________________________________________________________________________<br>Professional FreeSWITCH Consulting Services: <br><a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a><br><a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br><br>Official FreeSWITCH Sites<br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br><a href="http://confluence.freeswitch.org" target="_blank">http://confluence.freeswitch.org</a><br><a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br><br>FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a 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