[Freeswitch-users] Another FreeSWITCH First!

Peter J. Zandvoort peter at cindyandpeter.com
Wed Apr 1 10:58:57 PDT 2009


Excellent stuff Anthony! J

 

SIP over SMTP could actually be useful in a push-to-talk type of scenario.
Put the voice packets in an attachment. A slight delay, perhaps, but nicely
encapsulated in a totally standard protocol.

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nik
Middleton
Sent: Wednesday, April 01, 2009 12:08 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Another FreeSWITCH First!

 

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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