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<P><SPAN class=885140812-28022006>I have an issue about packetization/frame
length customization for the RTP stream generation.</SPAN></P>
<P><SPAN class=885140812-28022006>I needed to tune up the hardcoded
packetization value of Asterisk (which is 20ms). Consequently I applied the
patch located at <A
href="http://bugs.digium.com/view.php?id=5162">http://bugs.digium.com/view.php?id=5162</A>.</SPAN></P>
<P><SPAN class=885140812-28022006>Patch worked and I am now able to customize
packetization value per user/peer basis. However,</SPAN></P>
<UL>
<LI><SPAN class=885140812-28022006>packetization parameter setting has no
effect if used at the [default] part of sip.conf, <U>am I
correct?</U></SPAN></LI></UL>
<P><SPAN class=885140812-28022006>I performed several tests with SIP clients
from Grandstream. Please note the below;</SPAN></P>
<UL>
<LI><SPAN class=885140812-28022006>models are HT & GXP with the latest
firmware releases</SPAN></LI>
<LI><SPAN class=885140812-28022006>Asterisk version is 1.2.4</SPAN></LI>
<LI><SPAN class=885140812-28022006>choice of codec is G.729</SPAN></LI>
<LI><SPAN class=885140812-28022006>configured frame length is 50ms at the
clients and Asterisk</SPAN></LI>
<LI><SPAN class=885140812-28022006>clients register to Asterisk with
<EM>reinvite=no </EM>option for RTP proxying.</SPAN></LI></UL>
<P><SPAN class=885140812-28022006>Grandstream clients <U>fail</U> to send RTP
stream with 50ms customized frame rate. Instead, they send with the default
frame length which is 20ms. However, Asterisk sends with 50ms, as configured. As
a result, RTP flows with asymmetric bandwidth values, that is; 37Kbps from the
client and 20.5Kbps from the Asterisk.</SPAN></P>
<P><SPAN class=885140812-28022006>I trunked two Asterisk systems using SIP and
noticed that RTP flows perfectly, with </SPAN><SPAN
class=885140812-28022006>50ms/50ms frame lengths to/from any
Asterisk.</SPAN></P>
<UL>
<LI><SPAN class=885140812-28022006>When I checked the signaling with the
packet sniffer, I noticed that Grandstream implements <EM>ptime</EM> attribute
in SDP and explicitly declares frame length in the INVITE msg.
However, 200 OK msg from Asterisk does not include the confirmation of
the ptime value, resulting in Grandstream's decision to default to 20ms for
the frame length. <U>Is this a logical explanation?</U></SPAN></LI></UL>
<P><SPAN class=885140812-28022006><EM>ptime</EM> attribute of SDP is defined in
RFC2327 and is obviously an optional attribute. <U>What do you think about
implementing this option? Is it possible that interop problems may arise
due to the above phenomenon?</U></SPAN></P>
<P><SPAN class=885140812-28022006>Regards,</SPAN></P></DIV></BODY></HTML>