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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Well you almost had me there, but SIP over
SMTP? That was too much. <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Regards,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
freeswitch-users-bounces@lists.freeswitch.org
[mailto:freeswitch-users-bounces@lists.freeswitch.org] <b><span
style='font-weight:bold'>On Behalf Of </span></b>Anthony Minessale<br>
<b><span style='font-weight:bold'>Sent:</span></b> 01 April 2009 16:31<br>
<b><span style='font-weight:bold'>To:</span></b> Freeswitch-users<br>
<b><span style='font-weight:bold'>Subject:</span></b> [Freeswitch-users]
Another FreeSWITCH First!</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p><font size=3 face="Times New Roman"><span style='font-size:12.0pt'>The
FreeSWITCH team is excited to announce that FreeSWITCH is the first telephony
application to support the new SIP 4.1 protocol specification.<o:p></o:p></span></font></p>
<p><font size=3 face="Times New Roman"><span style='font-size:12.0pt'>Unlike
its predecessors, SIP 4.1 has been created with the collaboration of both the
jabber foundation and the IETF. With this match made in heaven, one can
now encapsulate an xml representation of a sip message, which in turn can
encapsulate a standard SIP 2.0 message making it possible to do more than ever
before.<br>
Other exciting features include: <o:p></o:p></span></font></p>
<p><font size=3 face="Times New Roman"><span style='font-size:12.0pt'>*) RFC
4109 support: (SIP over SMTP), allowing packets to traverse NAT with ease. <o:p></o:p></span></font></p>
<p><font size=3 face="Times New Roman"><span style='font-size:12.0pt'>*) Full
circle presence: endpoints must subscribe to each character in the printable
ASCII range that may be used to indicate presence and the server will send an
xml notification to the client for each character that is enabled whenever a
call takes place which in turn can be used to build a SIP 4.1 FYI packet that
can be sent to all the neighboring SIP devices so they may send themselves a
NOTIFY telling them that the light should blink if the same packet happens to
be sent from a neighbor. Then when the neighbor wants to send a presence
packet it establishes a dialog with the Third Party Presence Agent TPPA and
leaves the message there. Then it sends the server a PRESENCE packet,
which is then, relayed to the subscribers with the TPPA id so all the
subscribers can connect to the TPPA server to make the little light blink. <o:p></o:p></span></font></p>
<p><font size=3 face="Times New Roman"><span style='font-size:12.0pt'>*)
Retirement of SDP: SDP is deprecated in favor of a list of URL’s
describing the desired codec. The UA can then request this URL and get
the full details of the media requirements. The media port is negotiated
through trial and error where the calling UA asks the called UA if the port it
has guessed randomly is correct via direct TCP connection and an exchange of
XML PORT MARKUP LANGUGE XPML<o:p></o:p></span></font></p>
<p><font size=3 face="Times New Roman"><span style='font-size:12.0pt'>INVITE <a
href="mailto:bob@alice.com">bob@alice.com</a> SIP 4.1<br>
Content-type: sip-xml-encapsulated<br>
<SIP version=”4.1”><br>
<content type=”SIP-INVITE”><br>
<INVITE recipient=”<a
href="mailto:bob@alice.com">bob@alice.com</a>”><br>
<data type=”sip-2/0”/><br>
<![CDATA[INVITE <a
href="mailto:bob@alice.com">bob@alice.com</a> SIP 2.0<br>
To: <a href="mailto:bob@alice.com">bob@alice.com</a><br>
From: <a href="mailto:alice@bob.com">alice@bob.com</a><br>
Subject: SIP Rocks<br>
]]><br>
</data><br>
</INVITE><br>
</content> <br>
</SIP> <o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><br clear=all>
<br>
-- <br>
Anthony Minessale II<br>
<br>
FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>
ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
<br>
AIM: anthm<br>
<a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>
GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net">irc.freenode.net</a> #freeswitch<br>
<br>
FreeSWITCH Developer Conference<br>
<a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br>
<a href="http://iax:guest@conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</a><br>
<a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>
pstn:213-799-1400<o:p></o:p></span></font></p>
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