[Freeswitch-users] Another FreeSWITCH First!

[Raul Fragoso]

Agreed 100% !

That means we are all closer on taking 'mail-agents' to the holy-grail
level of voice communications !
I wonder if SIP 4.1 UAS will also handle MX records ? That would be
awesome ! I can't wait until we see something like mod_audio_spammer in
FreeSWITCH, so those lovely marketing workers can give voice to their so
much acclaimed phallic products.

Regards,

Raul

On Wed, 2009-04-01 at 13:58 -0400, Peter J. Zandvoort wrote:
> 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
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
> 
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> 
> 
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