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<DIV dir=ltr align=left><SPAN class=885140812-28022006>I have an issue about
packetization/frame length customization for the RTP stream generation.<SPAN
class=219350318-28022006><FONT face=Arial color=#0000ff
size=2> </FONT></SPAN></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=885140812-28022006><SPAN
class=219350318-28022006><FONT face=Arial color=#0000ff
size=2> </FONT></SPAN></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=885140812-28022006><SPAN
class=219350318-28022006> </SPAN>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
class=219350318-28022006><FONT face=Arial color=#0000ff
size=2> </FONT></SPAN></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=885140812-28022006><SPAN
class=219350318-28022006></SPAN></SPAN> </DIV>
<DIV dir=ltr align=left><SPAN class=885140812-28022006><SPAN
class=219350318-28022006><FONT face=Arial color=#0000ff
size=2> </FONT></SPAN>Patch worked and I am now able to customize
packetization value per user/peer basis. However,</SPAN></DIV>
<BLOCKQUOTE dir=ltr style="MARGIN-RIGHT: 0px">
<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
class=219350318-28022006><FONT face=Arial color=#0000ff
size=2> </FONT></SPAN></SPAN><SPAN class=885140812-28022006><SPAN
class=219350318-28022006> </SPAN></SPAN><SPAN
class=885140812-28022006>I performed several tests with SIP clients from
Grandstream. Please note the below;</SPAN>
<LI><SPAN class=885140812-28022006>models are HT & GXP with the latest
firmware releases</SPAN>
<LI><SPAN class=885140812-28022006>Asterisk version is 1.2.4</SPAN>
<LI><SPAN class=885140812-28022006>choice of codec is G.729</SPAN>
<LI><SPAN class=885140812-28022006>configured frame length is 50ms at the
clients and Asterisk</SPAN>
<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></BLOCKQUOTE></BODY></HTML>