<div dir="ltr"><div><div><div>In FS document of media proxy mode: <br> > FreeSWITCH has no control or even understanding of other SDP parameters.<br></div>Look like I have to find another way, like writing a custom module listening on specific event.<br></div>Any suggest?<br><br></div><div>Thanks to all of you.<br></div><div>rgds,<br></div><div class="gmail_extra"><br clear="all"><div><div class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div>Loi Dang Thanh<br></div>Phone : 841224.735.448<br></div>Email : <a href="mailto:loi.dangthanh@gmail.com" target="_blank">loi.dangthanh@gmail.com</a><br></div></div></div></div></div>
<br><div class="gmail_quote">On Wed, Nov 23, 2016 at 10:28 AM, Ken Rice <span dir="ltr"><<a href="mailto:krice@freeswitch.org" target="_blank">krice@freeswitch.org</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div link="blue" vlink="purple" lang="EN-US"><div class="m_9071376374511305464WordSection1"><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<u></u><u></u></span></p><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri",sans-serif"><u></u> <u></u></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"> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@<wbr>lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-<wbr>bounces@lists.freeswitch.org</a>] <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 <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.<wbr>freeswitch.org</a>><br><b>Subject:</b> Re: [Freeswitch-users] absolute_codec_string not working<u></u><u></u></span></p><div><div class="h5"><p class="MsoNormal"><u></u> <u></u></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.<u></u><u></u></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.<u></u><u></u></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?<u></u><u></u></p><div><p class="MsoNormal"><br clear="all"><u></u><u></u></p><div><div><div><div><div><div><div><p class="MsoNormal">Loi Dang Thanh<u></u><u></u></p></div><p class="MsoNormal">Phone : 01224.735.448<u></u><u></u></p></div><p class="MsoNormal">Email : <a href="mailto:loi.dangthanh@gmail.com" target="_blank">loi.dangthanh@gmail.com</a><u></u><u></u></p></div></div></div></div></div><p class="MsoNormal"><u></u> <u></u></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:<u></u><u></u></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.<u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></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:<u></u><u></u></p></div><p class="MsoNormal"><u></u> <u></u></p></div></div><div><div><div><div><div><div><div><p class="MsoNormal"><u></u> <u></u></p></div><p class="MsoNormal">Hi List, I got some trouble with using `absolute_codec_string` param.<u></u><u></u></p></div><p class="MsoNormal">My call scenario is pretty simple: caller <--> FS <--> callee.<u></u><u></u></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=<wbr>PCMU,GSM}sofia/gateway/callee/<wbr>$1"/>` in the dialplan.<u></u><u></u></p><div><p class="MsoNormal">But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the callee.<u></u><u></u></p></div><div><p class="MsoNormal" style="margin-bottom:12.0pt">not sure what I'm missing, helps would be appreciated.<u></u><u></u></p></div><div><p class="MsoNormal">Note that when I'm using `originate` application in fs_cli, things are good.<u></u><u></u></p></div><div><div><div><div><p class="MsoNormal">`originate {absolute_codec_string=PCMU}<wbr>sofia/gateway/caller/100 &bridge({absolute_codec_<wbr>string=PCMA}sofia/gateway/<wbr>callee/100`. <u></u><u></u></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.<u></u><u></u></p></div><div><p class="MsoNormal">rgds,<br clear="all"><u></u><u></u></p></div><div><div><div><div><div><div><div><div><p class="MsoNormal">Loi Dang Thanh<u></u><u></u></p></div><p class="MsoNormal">Phone : 84.1224.735.448<u></u><u></u></p></div><p class="MsoNormal">Email : <a href="mailto:loi.dangthanh@gmail.com" 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