<html xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns="http://www.w3.org/TR/REC-html40">
<head>
<meta http-equiv=Content-Type content="text/html; charset=us-ascii">
<meta name=Generator content="Microsoft Word 11 (filtered medium)">
<style>
<!--
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
        {margin:0cm;
        margin-bottom:.0001pt;
        font-size:12.0pt;
        font-family:"Times New Roman";}
a:link, span.MsoHyperlink
        {color:blue;
        text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
        {color:purple;
        text-decoration:underline;}
span.EmailStyle17
        {mso-style-type:personal-compose;
        font-family:Arial;
        color:windowtext;}
@page Section1
        {size:612.0pt 792.0pt;
        margin:72.0pt 90.0pt 72.0pt 90.0pt;}
div.Section1
        {page:Section1;}
-->
</style>
</head>
<body lang=EN-US link=blue vlink=purple>
<div class=Section1>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I’m trying to control the payload size of
conversations. Default seems to be 20 milliseconds for G.729A, but for regular
conversations I’ve been able to change this to 40 milliseconds. This lags
conversations somewhat but bandwidth usage is around 10kbits lower. I do this
by modifying the call to ast_smoother_new() on line 1225 in rtp.c:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> case AST_FORMAT_G729A:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>
if (!rtp->smoother) {<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> /*
rtp->smoother = ast_smoother_new(20) */<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:72.0pt;text-indent:36.0pt'><font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'>rtp->smoother =
ast_smoother_new(40);<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>This has no effect on the traffic between two
Asterisk-servers in a bridged conversation. The following is a crude drawing of
a G.729A SIP conversation. Both Asterisk servers have G.729A codec installed so
they’re able to bridge this codec and the client sends voice in G.729A:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> <o:p></o:p></span></font></p>
<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>
45
bytes packets<o:p></o:p></span></font></p>
<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>
Depending on client
35 kbit<o:p></o:p></span></font></p>
<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>
</span></font><font size=2 face="Courier New"><span lang=NO-BOK
style='font-size:10.0pt;font-family:"Courier New"'>------------------->
-----------------><o:p></o:p></span></font></p>
<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
lang=NO-BOK style='font-size:10.0pt;font-family:"Courier New"'>Client
Asterisk
Asterisk<o:p></o:p></span></font></p>
<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
lang=NO-BOK style='font-size:10.0pt;font-family:"Courier New"'>
<-------------------
<-----------------<o:p></o:p></span></font></p>
<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>
ast_smoother_new(40) 45 bytes packets<o:p></o:p></span></font></p>
<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'> 52
bytes packets
35 kbit<o:p></o:p></span></font></p>
<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'>
19 kbit<o:p></o:p></span></font></p>
<p class=MsoNormal style='text-autospace:none'><font size=2 face="Courier New"><span
style='font-size:10.0pt;font-family:"Courier New"'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='text-autospace:none'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'>Following is CLI printout from the Asterisk
server which the client registers with:<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Courier New"><span style='font-size:10.0pt;
font-family:"Courier New"'> == Spawn extension (context, 1000, 1) exited
non-zero on 'SIP/client-e097'<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Courier New"><span style='font-size:10.0pt;
font-family:"Courier New"'> -- Executing Dial("SIP/client-6b02",
"SIP/asterisk2/1000") in new stack<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Courier New"><span style='font-size:10.0pt;
font-family:"Courier New"'> -- Called asterisk2/1000<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Courier New"><span style='font-size:10.0pt;
font-family:"Courier New"'> -- SIP/asterisk2-559b answered
SIP/client-6b02<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face="Courier New"><span style='font-size:10.0pt;
font-family:"Courier New"'> -- Attempting native bridge of
SIP/client-6b02 and SIP/asterisk2-559b<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>What I want is to control the payload of the traffic between
the Asterisk servers. Reinvites is not allowed in this particular system.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Best regards,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Alf Thomas Nilsen<o:p></o:p></span></font></p>
</div>
</body>
</html>