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@alice.com SIP 4.1 Content-type: sip-xml-encapsulated <SIP version=”4.1”> <content type=”SIP-INVITE”> <INVITE recipient=”bob@alice.com”> <data type=”sip-2/0”/> <![CDATA[INVITE bob@alice.com SIP 2.0 To: bob@alice.com From: alice@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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Well you almost had me there, but SIP over SMTP? That was too much. Regards, ________________________________ From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@alice.com SIP 4.1 Content-type: sip-xml-encapsulated <SIP version="4.1"> <content type="SIP-INVITE"> <INVITE recipient="bob@alice.com"> <data type="sip-2/0"/> <![CDATA[INVITE bob@alice.com SIP 2.0 To: bob@alice.com From: alice@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@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
You know you could write a transport plugin for Sofia that would do SIP over SMTP :P /b On Apr 1, 2009, at 11:07 AM, Nik Middleton wrote:
Well you almost had me there, but SIP over SMTP? That was too much.
Regards,
Brian West brian@freeswitch.org -- Meet us a ClueCon! http://www.cluecon.com
Here's a sample SIP/SMTP INVITE (responses omitted for clarity) MAIL FROM: <dave@3c.co.uk> RCPT TO: <marina@3c.co.uk> DATA Call me . --Dave Sent from my iPhone On 1 Apr 2009, at 09:15, Brian West <brian@freeswitch.org> wrote:
You know you could write a transport plugin for Sofia that would do SIP over SMTP :P
/b
On Apr 1, 2009, at 11:07 AM, Nik Middleton wrote:
Well you almost had me there, but SIP over SMTP? That was too much.
Regards,
Brian West brian@freeswitch.org
-- Meet us a ClueCon! http://www.cluecon.com
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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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Nik Middleton Sent: Wednesday, April 01, 2009 12:08 PM To: freeswitch-users@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@alice.com SIP 4.1 Content-type: sip-xml-encapsulated <SIP version="4.1"> <content type="SIP-INVITE"> <INVITE recipient="bob@alice.com"> <data type="sip-2/0"/> <![CDATA[INVITE bob@alice.com SIP 2.0 To: bob@alice.com From: alice@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@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Nik Middleton Sent: Wednesday, April 01, 2009 12:08 PM To: freeswitch-users@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@lists.freeswitch.org [mailto:freeswitch-users-bounces@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@alice.com SIP 4.1 Content-type: sip-xml-encapsulated <SIP version=”4.1”> <content type=”SIP-INVITE”> <INVITE recipient=”bob@alice.com”> <data type=”sip-2/0”/> <![CDATA[INVITE bob@alice.com SIP 2.0 To: bob@alice.com From: alice@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@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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) http://TwiTR.Me (Cross Social Network Messaging Bus) http://TweetOnTV.net (Private Label Social TV Platform) http://TwebEX.com (Twitter Based Online Web Conference Platform) http://TwitrShare.com (Send Picture and Message to Tweet Contacts) http://Twookups.com (Twitter Matching Service) http://TweetUp.ws (Twitter based MeetUp service) 2009/4/1 Anthony Minessale <anthony.minessale@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@alice.com SIP 4.1 Content-type: sip-xml-encapsulated <SIP version=”4.1”> <content type=”SIP-INVITE”> <INVITE recipient=”bob@alice.com”> <data type=”sip-2/0”/> <![CDATA[INVITE bob@alice.com SIP 2.0 To: bob@alice.com From: alice@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@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
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participants (7)
-
Anthony Minessale
-
Brian West
-
David Knell
-
EdPimentl
-
Nik Middleton
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Peter J. Zandvoort
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Raul Fragoso