I have tried to apply the patch attached to bug id 5162, but the behavior appears to be the same (I'm using sip channels). When doing an RTP debug I still see g.729 packets with len=20 (not 40) and g.723 packets with len=24 (not 48). I tried posting my findings to the bug log, but I'm unable to get a login name so I'm posting here. Is there any configuration that needs to happen besides putting "packetization = xx" in the peer definitions?
<br>Also, any reason why there is no definition for g.723 in rtp_codec_table RTP_CODEC_TABLE[]?<br>Thanks,<br><br>-fernando<br><br><div><span class="gmail_quote">On 1/21/06, <b class="gmail_sendername">Dan Austin</b> <
<a href="mailto:Dan_Austin@phoenix.com">Dan_Austin@phoenix.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">Bug id 5162 makes it possible, at least for G729. It
should apply to 1.2.X</font></span></div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">It hasn't been reviewed or accepted into trunk, but works
very well for us.</font></span></div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2"></font></span> </div>
<div dir="ltr" align="left"><span><font color="#0000ff" face="Arial" size="2">Dan</font></span></div><br>
<blockquote style="margin-right: 0px;">
<div dir="ltr" align="left" lang="en-us">
<hr>
<font face="Tahoma" size="2"><b>From:</b> <a href="mailto:asterisk-dev-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-dev-bounces@lists.digium.com</a>
[mailto:<a href="mailto:asterisk-dev-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-dev-bounces@lists.digium.com</a>] <b>On Behalf Of </b>Fernando
Lombardo<br><b>Sent:</b> Saturday, January 21, 2006 7:05 AM<br><b>To:</b>
<a href="mailto:asterisk-dev@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-dev@lists.digium.com</a><br><b>Subject:</b> [asterisk-dev] frame
size<br></font><br></div>
<div></div><span class="gmail_quote"></span><span><span class="gmail_quote"></span>Hello, I've been experimenting with Asterisk and
transcoding. I have a question. Is there a way to make it so that the RTP is
sent in 40 byte frames (for g.729) or 48 byte frames (for g.723).I noticed
when doing a rtp debug in Asterisk that it is actually sending 20 byte frames
for both g729/g723 while the other end sends 40 bytes/48bytes respectively.
This would cause asymmetric bandwidth consumption. Thanks, <br clear="all"><br clear="all">-fernando<br></span></blockquote>
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</a><br><br><br></blockquote></div><br><br clear="all"><br>-- <br>-fernando