[Freeswitch-users] Another FreeSWITCH First!

EdPimentl edpimentl at gmail.com
Wed Apr 1 09:53:15 PDT 2009


LOVE!!!!!
Now we can create Twitter-Voip apps....

Best regards,
-E
CEO and Founder
Gpro.ws
edpimentl [SKype | GoogleTalk | Twitter ]
http://Twitter.com/edpimentl

http://AskTwitR.com   (Real Time Twitter Search & Reputation Management)
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http://TweetUp.ws      (Twitter based  MeetUp service)


2009/4/1 Anthony Minessale <anthony.minessale at gmail.com>

> 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 <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
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>
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