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 From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of L?i Ð?ng Sent: Tuesday, November 22, 2016 9:07 PM To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] absolute_codec_string not working Hi @Michael, you were right, I'm intentionally using media_proxy for FS, since I want to reduce CPU usage on FS machine. In this case, I just want to limit the codecs used for each endpoint, and codec negotiation will be handled by them. e.g: caller use PCMA, PCMU, GSM by its own in INVITE, I want to limit the callee to only use PCMA,GSM. Look like `absolute_codec_string` is not what I'm looking for right? Any way out? Loi Dang Thanh Phone : 01224.735.448 Email : loi.dangthanh@gmail.com On Tue, Nov 22, 2016 at 9:57 PM, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com> > wrote: using proxy_media is my best guess but can’t tell with this little info. On Nov 22, 2016, at 5:27 AM, Lợi Đặng <loi.dangthanh@gmail.com <mailto:loi.dangthanh@gmail.com> > wrote: Hi List, I got some trouble with using `absolute_codec_string` param. My call scenario is pretty simple: caller <--> FS <--> callee. 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. But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the callee. not sure what I'm missing, helps would be appreciated. Note that when I'm using `originate` application in fs_cli, things are good. `originate {absolute_codec_string=PCMU}sofia/gateway/caller/100 &bridge({absolute_codec_string=PCMA}sofia/gateway/callee/100`. 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. rgds, Loi Dang Thanh Phone : 84.1224.735.448 Email : loi.dangthanh@gmail.com <mailto:loi.dangthanh@gmail.com> _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org