[Freeswitch-users] Another FreeSWITCH First!

[Nik Middleton]

Well you almost had me there, but SIP over SMTP?  That was too much. 

 

Regards,

 

________________________________

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 01 April 2009 16:31
To: Freeswitch-users
Subject: [Freeswitch-users] Another FreeSWITCH First!

 

The FreeSWITCH team is excited to announce that FreeSWITCH is the first
telephony application to support the new SIP 4.1 protocol specification.

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.
Other exciting features include: 

*) RFC 4109 support: (SIP over SMTP), allowing packets to traverse NAT
with ease. 

*) 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. 

*) 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

INVITE bob at alice.com SIP 4.1
Content-type: sip-xml-encapsulated
<SIP version="4.1">
  <content type="SIP-INVITE">
    <INVITE recipient="bob at alice.com">
      <data type="sip-2/0"/>
      <![CDATA[INVITE bob at alice.com SIP 2.0
To: bob at alice.com
From: alice at bob.com
Subject: SIP Rocks
]]>
      </data>
    </INVITE>
  </content>  
</SIP> 



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale at hotmail.com
<mailto:MSN%3Aanthony_minessale at hotmail.com> 
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
<mailto:PAYPAL%3Aanthony.minessale at gmail.com> 
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
<mailto:sip%3A888 at conference.freeswitch.org> 
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org> 
pstn:213-799-1400

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