<div dir="ltr"><div>Hi @Michael, you were right, I&#39;m intentionally using media_proxy for FS, since I want to reduce CPU usage on FS machine.<br><br></div><div>In this case, I just want to limit the codecs used for each endpoint, and codec negotiation will be handled by them.<br></div>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&#39;m looking for right? Any way out?<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 : 01224.735.448<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 Tue, Nov 22, 2016 at 9:57 PM, Michael Jerris <span dir="ltr">&lt;<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div style="word-wrap:break-word">using proxy_media is my best guess but can’t tell with this little info.<div><br><div><blockquote type="cite"><div><div class="h5"><div>On Nov 22, 2016, at 5:27 AM, Lợi Đặng &lt;<a href="mailto:loi.dangthanh@gmail.com" target="_blank">loi.dangthanh@gmail.com</a>&gt; wrote:</div><br class="m_7846363809754745801Apple-interchange-newline"></div></div><div><div><div class="h5"><div dir="ltr"><div><div><div><br></div>Hi List, I got some trouble with using `absolute_codec_string` param.<br></div>My call scenario is pretty simple: caller &lt;--&gt; FS &lt;--&gt; callee.<br></div>My caller compose `m=audio 7078 RTP/AVP 8 0 101` in its INVITE, and I&#39;m doing `&lt;action application=&quot;bridge&quot; data=&quot;{absolute_codec_string=<wbr>PCMU,GSM}sofia/gateway/callee/<wbr>$1&quot;/&gt;` in the dialplan.<br><div>But FS still use `m=audio 22952 RTP/AVP 8 0 101` in the INVITE to the callee.<br></div><div>not sure what I&#39;m missing, helps would be appreciated.<br><br></div><div>Note that when I&#39;m using `originate` application in fs_cli, things are good.<br></div><div><div><div><div>`originate {absolute_codec_string=PCMU}<wbr>sofia/gateway/caller/100 &amp;bridge({absolute_codec_<wbr>string=PCMA}sofia/gateway/<wbr>callee/100`. <br></div><div>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.<br><br></div><div>rgds,<br clear="all"></div><div><div><div class="m_7846363809754745801gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div>Loi Dang Thanh<br></div>Phone : 84.1224.735.448<br></div>Email : <a href="mailto:loi.dangthanh@gmail.com" target="_blank">loi.dangthanh@gmail.com</a><br></div></div></div></div></div>
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